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Hi,
Mini-PCI cards such as those from Junghanns (BRI or E1,
http://www.junghanns.net/en/produkte.html), are available.
Are these modules really usable as it doesn't seem very easy to integrate
those in appliances ?
Can someone share its experience with those ?
More specifically, how can you add
Morning! Thank you very much for the answers. These are my considerations:
If it's a pre-paid app and you're doing a Dial command (like a calling
card), why not use the limit (L) feature that's built in?
Because limit the time before the call starts does not cover all the
possible cases of
Hello Philippe!
Would you by any chance have asterisk running with gtalk? I saw your mail
there. If so perhaps we could test. Because all others I have found either
don't have gtalk, so we tried jingle, which was still a bit problematic or if
they had pure gtalk, they weren't really upto the
Hi Julien,
The Gtalk call to your buddy fails because of a mismatch in the UDP
ports for RTP. Try to disable the 'strictrtp' option in your rtp.conf
file. Question : did you scramble the IP addresses?
The Jingle call fails because of Google's XMPP network refusing to
relay jingle packets wrapped
Hello Philippe!
I've set the strictrtp to no. Where should I have scrambeld my address? I'm
not aware of it. The only thing that maybe is that I'm on a small LAN behind
the firewall. May this be relevant?
About the bugtracker. I've had a look there, but couldn't really find the
place
Hi Julien,
I just placed a call from my GoogleTalk account to your Asterisk
server, reached your voicemail (or at least I guess cause the welcome
message is in German), and left a message.
Cheers,
Philippe
On Tue, Oct 28, 2008 at 10:24 AM, Julien Claassen [EMAIL PROTECTED] wrote:
Hello
Hi Daniel;
1) What the SIP client that Nokia has it other than fring?
2) Coming back to open vpn: can open vpn cient work on Nokia? Do u have any
idea about this?
If to talk about siphone, I did not test it, but does it work with Nokia?
Regards
Bilal
[...]
What about Nokia Communicator? Any
Gordon Henderson wrote:
On Sat, 25 Oct 2008, Joseph L. Casale wrote:
X100P.
Yeah I saw these but they are single port and I need at least 2 ports. I
only have 1 free pci slot as well.
OpenVox.
Gordon
I don't know if this is still the case, but nxtvox used to be much
cheaper,
Hello All,
I'am a new Asterisk user, and i have the following question.
The following is the Status of all open channels from my Asterisk
system, which was received through the
Asterisk Manager Interface ((AMI)).
action:
Simith,
normaly, the caller channel have a minor uniqueid.
Simith Nambiar escreveu:
Hello All,
I'am a new Asterisk user, and i have the following question.
The following is the Status of all open channels from my Asterisk
system, which was received through the
Asterisk Manager
Just built myself a little test server with an Atom 230 processor in it
and am quite impressed with it so-far.
Wondering is anyones used one in anger for a VoIP platform?
I'm after something with a bit more oomph than the VIAs I'm currently
using that I can use in a small box (mini ATX size)
On Tue, Oct 28, 2008 at 12:30 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
Just built myself a little test server with an Atom 230 processor in it
and am quite impressed with it so-far.
Wondering is anyones used one in anger for a VoIP platform?
I'm after something with a bit more oomph
Hi,
I've ran into an issue with a PRI provider in a major metropolitan area that
I haven't needed to deal with before and I was hoping someone might have
some insight on how to handle this within the Asterisk dialplan.
At this location users can't always tell if a number is long distance or not
Steven's writeup is great and explains a lot of this.
A shorter answer from a different perspective:
The folks that devloped the fax V.protocols took into acount typical
copper problems like noise or echo. But what they never conceived of as
even being possible is that a call might shift
Hola Victor,
Remember that you could use an option from the 'Dial' command itself. If
you examine the Dial command documentation you will this option:
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
On October 28, 2008 12:58:25 pm JD wrote:
The folks that devloped the fax V.protocols took into acount typical
copper problems like noise or echo. But what they never conceived of as
even being possible is that a call might shift around in the time
domain. Thanks to jitter/latency, the delay
On Tue, 28 Oct 2008, Steve Totaro wrote:
On Tue, Oct 28, 2008 at 12:30 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
Just built myself a little test server with an Atom 230 processor in it
and am quite impressed with it so-far.
Wondering is anyones used one in anger for a VoIP platform?
On Tue, 28 Oct 2008, JD wrote:
Steven's writeup is great and explains a lot of this.
A shorter answer from a different perspective:
The folks that devloped the fax V.protocols took into acount typical
copper problems like noise or echo. But what they never conceived of as
even being
On Tue, Oct 28, 2008 at 1:17 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Tue, 28 Oct 2008, Steve Totaro wrote:
On Tue, Oct 28, 2008 at 12:30 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
Just built myself a little test server with an Atom 230 processor in it
and am quite impressed with
I am using a PBX in my house built on the Intel Atom 330 dual core.
Here is a review of it.
http://www.neoseeker.com/Articles/Hardware/Reviews/D945GCLF2_atom_330/
The only problem I had was the network card driver built into CentOS 5.2
didn't recognize the correct model of the card. After I
On Tue, 28 Oct 2008, Steve Totaro wrote:
I will never install anything called Linpus, lol, that might be
worst name for a distro I have ever heard.
I know!
I am watching some Fedora and CentOS threads, I want everything to
work, last I checked, the onboard video cam and the wireless NIC was
copper has nothing to do with it. You're talking about conventional PSTN
(circuit
switched) technologies.
Even more succinctly, it is the difference between streaming and packetized
data. Circuit switched isn't even a necessary qualifier. Dedicated bandwidth
with minimal buffering is
Gordon Henderson wrote:
snip /
Damn - I've just found it in the UK too:
http://www.lambda-tek.com/componentshop/index.pl?prodID=1606310
Must resisst .
I just wish there was a fanless version - one feature which I like in the
VIA boards I use.
Wow, that's an amazing
Hi All,
I've looked through the archives and tried several variations in Google, and I
haven't found anything on-point... So I'm hoping someone here may be able to
help this relative Asterisk neophyte shed some light on an issue:
I have a box running Asterisk 1.4.22 in our lab with several
Gordon Henderson wrote:
but it's very do-able, given good Internet connections.
[...]
I think your statements were just a bit too strong - I agree
wholeheartedly about the V. protocols and copper, but I've found in
practice that faxing over IP is not just theoretically possible, but quite
On Tue, 28 Oct 2008, Alan Lord wrote:
I just wish there was a fanless version - one feature which I like in the
VIA boards I use.
Wow, that's an amazing price for the mobo. Though, like you, WTF do
Intel insist on using a chipset that needs fan cooling and draws about 4
times as much
Gordon Henderson wrote:
[snip]
That's a prety good price - I didn't think the dual-core Atoms were out
yet. I'll wait until I can get something in the UK..
Damn - I've just found it in thet UK too:
http://www.lambda-tek.com/componentshop/index.pl?prodID=1606310
Must
Lincoln King-Cliby wrote:
Hi All,
I've looked through the archives and tried several variations in Google, and
I haven't found anything on-point... So I'm hoping someone here may be able
to help this relative Asterisk neophyte shed some light on an issue:
I have a box running Asterisk
The folks that devloped the fax V.protocols took into acount typical
copper problems like noise or echo. But what they never conceived of as
even being possible is that a call might shift around in the time
domain. Thanks to jitter/latency, the delay time of a call can change in
the middle
This is caused by your DSL/cable modem buffering the data internally. To
resolve this you need to feed the modem at a data rate that prevents it from
buffering. I solved this by using a shaping bridge server that does nothing but
control the data in and out of the internet connection. I have
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP
it is mentioned MWI is now working.
In my testings with lastest 02123 firmware, MWI is blinking when
missed calls but
Olivier wrote:
2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults
On Oct 27, 2008, at 10:34 AM, Richard Brady wrote:
Hi folks
I have a handset talking to Asterisk, which in turn puts the call
through to an ITSP.
The handsets likes to send audio in 40ms frames (even though
Asterisk requests 20ms frames with a ptime header in the SDP).
The ITSP
Lincoln King-Cliby [EMAIL PROTECTED] writes:
Periodically I'm seeing calls placed from the 7961s through anything
on the PBX that requires digit entry (the Auto Attendant, Voicemail,
etc.) 'randomly' drop; extension-to-extension calls
extension-to-PSTN, and PSTN-to-extension calls never have
What is involved in provisioning Asterisk to use a multiline analog service
from our local telco?
I will only have one twisted pair entering in on a OpenVox card but am not sure
how Asterisk
interprets and deals with two incoming calls and/or two outgoing calls?
Thanks!
jlc
Gordon Henderson wrote:
I'd never say it was reliable enough to trust in a commercial setting, but
I think your statements were just a bit too strong - I agree
wholeheartedly about the V. protocols and copper, but I've found in
practice that faxing over IP is not just theoretically possible,
Hello guys, anybody here that can help me checking out this xml file, cause
I am traying to configure some cisco 7911G phones to asterisk and I can't
get it done
thanks
a paste of the file is here:
http://pastebin.ca/1239083
--
http://celord.blogspot.com/
Some additional information.
I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an unusual
result:
[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)
I'm not sure if it's the only issue but you're going to have issues with
phonelabelEtiqueta_del_telefono/phonelabel
The text within the phonelabel tag is a maximum of 11 or 12 characters (I
can't remember off the top of my head), if it's longer than that--I count 21
characters in the example,
When I send email from my local asterisk machine, my IP address get's
RBL'd.
Asterisk is my only reason for running sendmail, so to keep it simple, I
tried to make my ISP's mail server a 'smart host' (relaying to a trusted
mail server) but my ISP doesn't allow ANY kind of relaying these days.
Hi,
A newbie here trying to learn Asterisk. I've installed PiAF v.1.3(PBX in A
Flash) and trying to set up the TDM808E card as a test. For now I only have
one analog line. I went into the FreePBX interface and created a ZAP trunk
with 1 as the Zap Identifier.
When I try to call out, I
You need to implement SMTP-AUTH and log in when sending mail to your
smart host. I have a template for Postfix to do that. Many *nix distros
have Postfix with a sendmail compatible binary in front of it.
Lyle Giese
LCR Computer Services, Inc.
[EMAIL PROTECTED] wrote:
When I send email from my
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty useless.
They dont have any central management or provisioning, as well as a pretty bad
interface.
Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have
* Central Management for all the phones (We
Hi,
A lot of people talk about grooming to make VoIP work smoothly, not just
for FAX. However, most people can only achieve grooming in one
direction. Their ISP will not cooperate, and groom what is sent to the
subscriber. Unless you just keep your DSL link very lightly loaded, by
doing no
JD wrote:
Gordon Henderson wrote:
but it's very do-able, given good Internet connections.
[...]
I think your statements were just a bit too strong - I agree
wholeheartedly about the V. protocols and copper, but I've found in
practice that faxing over IP is not just
You can use Cisco phones as long as they have a SIP image.
Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty
useless.
They dont have any central management or provisioning, as well as a
pretty bad interface.
Can anyone reccomend any
Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty
useless.
They dont have any central management or provisioning, as well as a
pretty bad interface.
This is completely incorrect. Linksys SPA-942s *do* have the ability
for central
There are some decent third part central management systems for the
Linksys phones (the company I work for write one of them) - the same
piece of software will also manage Polycom and Snom phones (As well as
Aastra phones and a few others)
With regards to programmable buttons, the snoms are
I have been able to repeat the results at other locations. The location that
has 26 pages is a linksys PAP2T our accounting person uses remotely to fax
stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL line.
I can send faxes from my test sever at home that is using an
...as others have mentioned, yes they do have the ability to be centrally
managed, provisioned, configured.
Also, from the latest firmware, 6.x.x:
- Ability to use line buttons as quick dials
- Ability to query centralized LDAP for directory (I haven't tested
this one yet)
So, the
[EMAIL PROTECTED] wrote:
When I send email from my local asterisk machine, my IP address get's
RBL'd.
I use msmtp;
http://msmtp.sourceforge.net/
Here is my /etc/msmtprc
account default
host mail.bellsouth.net
auto_from on
maildomain bellsouth.net
syslog LOG_MAIL
we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line
we have for our office a different ADSL with one IP shared.
Two identical setup snom 360 (except the user name) with two public IP
addresses are connected at the hub to the server / DSL line
phone A can call B, B cannot
You can use the Cisco phones with either the SIP of the SCCP image.
Though I do agree that the SIP image is a bit easier to setup and auto-
provision, the SCCP image is a more native (obviously) implementation.
The chan-sccp-b project has nearly every feature usable on these
phones working
So I have (and have had) jabber configured for some time, specifically
for GTalk, but something has occurred to me. If somebody happens to
send an IM (text) to that account, nobody is going to be receiving it.
I'd like to send a canned message back to any sender of an IM.
Possible?
b.
Try using a R or r on the Dial command, the R option is better for you in my
opinion.
i.e Dial(Zap/G2/1${EXTEN},30,R) or Dial(Zap/G2/1${EXTEN}|30|R)
The R option is going to generate a ring tone when the callee indicates
ringing and is going wait for an Answer. As Progress is just for early
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