Andrew Kohlsmith wrote:
On Friday 04 November 2005 11:57, Matt wrote:
I have a Digium TE100 that I will be connecting to a T1. The T1 provider
is asking whether the T1 voice circuits are T1 analog or T1 digital.
?? T1 is digital. There is no analog.
I think what the provider is
Hello
I want to test asterisk with an H323 client. In Windows XP there is phone
dialer which can use H323. In Phone dialer I set H323 Line for phone calls
and Internet calls.
In Phone and Modem properties H323 provider I set:
H.323 gatekeeper: 192.168.0.20 (asterisk on my LAN)
Log on
I just wanted to let every know that My
complaint is not with Thirdlane. The Webmin module that they wrote is awesome
and is well worth
Getting. I have had a problem with a
person that took us for a loop no pun intended.
Alex has been a wonderful help and I would
defiantly suggest his
Hi,
What is the best way to allow a user to select the language they hear in
the dial plan? In other words I want the phone to answer Hello welcome
to ABC company to continue in English press 1 Followed by the same
thing in Spanish (Mexican Spanish - I live in the South West United
States)
I'm sorry, that previous message might have made more sense if it had
all the information that I had intended to send.
We are having moments of silence in the middle of phone calls.
Generally it's not more than a few seconds, but it's still a nuisance.
Our IAX providers (we have 2) become
In my FC2 box with a new-ish AMP install, AMP can be effectively disabled by
commenting out /usr/sbin/ampportal start in /etc/rc.d/rc.local
However, the ampportal script also calls safe_asterisk so you have to add
a reference to safe_asterisk in rc.local if you want Asterisk to start at
boot, or
Hi,
One of my clients wants to have a real person answer the phone (imagine
that). After hours we will have an automated operator answering the
phone, but several users will be using an application during the day
that requires an automated operator. How does one create a transfer in
the dial
I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
phones.
All phones register fine with * and I can place outbound calls with
no problem.
I can call from the X-Pro to any other X-Pro. I can call from UIP200
to any other X-Pro. However, the UIP200 cannot receive calls.
On Fri, Nov 04, 2005 at 09:34:29AM -0600, Eric ManxPower Wieling wrote:
Chris Bagnall wrote:
Hello all,
Is there a list of phones that reliably support SIP early dial? One of the
really nice things I've noticed about the 7960 (SCCP) is that each digit is
sent straight to asterisk, so when
We have also experienced momentary periods of silence in the middle of
phone calls.
I'm wondering if this could be related to the IAX peers becoming
unreachable?
Has anyone experienced moments of silence during a call, and do you know
what the causes might be?
My wife running out of
Hi,
Is there a way to play .gsm sound files on Windows. Is there an
extension for Windows Media Player or Real Player to allow playing of
these files?
Thanks
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Try putting a line at the very bottom of /etc/rc.d/rc.local like
/sbin/modprobe ztdummy
Moj
Min Hwan Chang wrote:
Greetings,
On Fedora Core 1 with a Generic X100P
For the life of me, I can't seem to get ztdummy to load on startup. I
go through the usual routine of uncommenting the '#' in
We had the same kind of issue with a VPN to our Calgary office. The VPN was
so slow, it was unusable and would periodically drop. Doing a traceroute, we
found that the packets would take a stupid route to Calgary, basically they
would be backhauled to routers all over North America, when we *knew
seems every 10 sec something is happeneing on your network...
make sure your router is rebooted often if you have QOS on it has they tend to get behind on queues..
or UDP crc checksum failing in router.. that happened to me
on a linksys
your ping is ok 60 is good
i would also test my lan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Clark
Sent: Saturday, 5 November 2005 1:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom IP 600/601 microbrowser specs
I've seen a couple of
Quicktime knows what to do with them.
On 11/4/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Is there a way to play .gsm sound files on Windows. Is there an
extension for Windows Media Player or Real Player to allow playing of
these files?
Thanks
you could wait infinitely or try users list..On 11/4/05, harry gaillac [EMAIL PROTECTED] wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :)
Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered
my bad you are.. lol didnt realize..
On 11/4/05, Jimmy Smith [EMAIL PROTECTED] wrote:
you could wait infinitely or try users list..On 11/4/05, harry gaillac
[EMAIL PROTECTED] wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean redirect host + port :)
Sip agents send sip
Hi,
There is a winamp plug-in, you can find it at:
http://www.winamp.com/plugins/details.php?id=142107
Regards,
Claudio
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If you use a SIP video phone with ASterisk. Does the monitor function
record video and audio?
Regards,
Chris___
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Is there a provider that has good support
and answers the phone? (=
I need to get lines for my Asterisk server
and want to move from broadvoice.com.
So far I havent been able to get
anyone on the phone.
Too funny..
Anyway. I want to work with a larger company
one that we know wont
Also, there's a WinAmp plugin to handle GSM files
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-BJ Weschke
-Sent: Friday, November 04, 2005 4:08 PM
-To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
-Non-Commercial Discussion
-Subject: Re:
Try calleveryone.com Yes.. I have blown their trumpet before. They
are a very good company with great support.
On 11/3/05, Jason Brashear [EMAIL PROTECTED] wrote:
Is there a provider that has good support and answers the phone? (=
I need to get lines for my Asterisk server and want to
Quicktime ? It's not an extension of either but it does GSM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Friday, November 04, 2005 23:43
To: Asterisk - Users
Subject: [Asterisk-Users] GSM sound player for windows?
Hi,
Is there a way
I understand that I
must pay for a support license to download Cisco firmware, so Im not
trying to pirate it. I simply want to know what I need to buy in order to get
firmware files for my phones. Does anyone have any helpful links they can give?
What does this license cost?
I was interested in getting feedback from current communities of Roomity.com and let you know the recent improvements we are working on for better interface.Roomity.com v 1.5 is a web 2.01/RiA poster child community webapp. This new version adds broadcast video, social networking such as favorite
Sorry, what exactly does your spam have to do with Asterisk Users?
--
Cheers,
Matt Riddell
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http://freevoip.gedameurope.com (Free Asterisk Voip Community)
I have not loaded the 7.1 firmware - it must have just recently been
released - it was not on their site last week, so I could not tell you,
but will load it up and play with it over the weekend.
I am running * CVS-HEAD with cnah_sccp-20050922 - my MWI is working as
well as service URL. Does the
Jimmy Smith wrote:
seems every 10 sec something is happeneing on your network...
make sure your router is rebooted often if you have QOS on it has they
tend to get behind on queues..
or UDP crc checksum failing in router.. that happened to me
on a linksys
your ping is ok 60 is good
i
You probably do not need firmware. I have tried several versions on
70s, 60s, 12s, 05s and 20s (not 02s) with success.
If they are not even looking for TFTP, then from the phone, hit
Settings-2**#, and erase. Make sure your DHCP server is kicking out
option 150 right (the correct TFTP server) -
Hello,
Rumor has it that the TDM2400 series cards will be available in
the next week or so. If you are a distributor that has pricing /
availability information, please contact me offlist. I am putting together
a solution for a client that will require a TDM2420E (8 Port FXS w/ Echo
Mexican Spanish..
Ha, funny term...
:)
Mexican Spanish = mx from MeXico...
es = ESpain...
So, es would be.. humm. Espain Spanish?
Chuck Bunn wrote:
Hi,
What is the best way to allow a user to select the language they hear in
the dial plan? In other words I want the phone to answer Hello
[EMAIL PROTECTED] wrote on 11/04/2005
04:34:18 PM:
Try calleveryone.com Yes.. I have blown their trumpet before.
They
are a very good company with great support.
Do they support IAX or just SIP? I've been reluctant
to use a SIP provider for a number of reasons, including difficulties in
es = ESpain...
So, es would be.. humm. Espain Spanish?
In Spain, Spain is España, so there's where the es comes from.
Espain doesn't exist. As you may have noticed, I'm spanish :)
Ruben
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Please Cancel this Post. I posted with an incorrect email address.
Thanks,
Chip
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Friday, November 04, 2005 2:04 PM
To: Chip Pleasants
Subject: Your message to Asterisk-Users awaits moderator approval
Your mail to
Hi,
I am wondering if it is possible to adapt different dial plan depending
upon which channel answered the call. I am pretty sure we can do this
by putting them in to two different context but I want to avoid
multiple context as far as possible.
Is it possible to know which channel answered
Jason Brashear wrote:
Is there a provider that has good support and answers the phone? (=
I need to get lines for my Asterisk server and want to move from
broadvoice.com.
So far I haven’t been able to get anyone on the phone.
Too funny…..
I was able to get them on the phone today but it
Kanuri, Seshu (Company IT) wrote:
Iain Barker Wrote:
-
Our experience with over 10 or more participants
in a single Asterisk conference was that quality
degraded quite rapidly.
Is this really true as there were many in this list
who had confirmed
That concept is not bad; except when the CEO from the same company as the
tech that calls all the time happens to call you from what appears to be the
same caller id, and the CEO ends up hearing rap or hard rock...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 11/04/2005 04:34:18 PM:
Try calleveryone.com Yes.. I have blown their trumpet before. They
are a very good company with great support.
Do they support IAX or just SIP? I've been reluctant to use a SIP
provider for a number of
I am wondering if it is possible to adapt different dial plan depending upon
which channel answered the
call. I am pretty sure we can do this by
putting them in to two different context but I want to avoid multiple context
as far as possible.
Is it possible to know which channel
We run SIP phones (polycom 501s) to * and a tdm40b for PSTN out, mostly
SIP - PSTN calls but a few SIP - SIP calls. We haven't implemented
any vlans or QoS so from time to time the network traffic and latency
get the better of our communication. Usually this manifests as a slight
crackle,
Hello users,
Somebody knows a good flash operator that works fine with unicall
channels? I don't know any one that can support this :-(
And you Steve Underwood, could you give me a tip?
Best regards,
Loureiro.
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED]
Steve:
That's exactly what I'm using. Incoming calls work like a charm
but when I try calling I get a protocol error. My provider says that
for outgoing I need to use fx signalling. I see that in unicall.conf
there's such a thing as protocolvariant=fx but if I uncomment that
line, unicall gives
Well, that's why you'd see languages refered to as
ES_mx
ES_es
ES_pe
ES_ve
And so on much better way to deal with language variants...
2005/11/4, Andres Tello Abrego [EMAIL PROTECTED]:
Mexican Spanish..Ha, funny term...:)Mexican Spanish = mx from MeXico...es = ESpain...So, es would be..
Intermittently Ill get
calls from my only SIP extension to itself via the Zap/1. I have no clue and
have found nothing online. I have listed my configurations and a sample of the
console messages I see why debugging. Right now it only happens to the 6000 extension.
Any assistance is
yes, it works!
But if set as such, we can not get callid, right?
And the callerid is a must element for us.
Any other advice ?
Thanks ,
Rich Adamson [EMAIL PROTECTED] 写道：
On Thursday 03 November 2005 02:50, Gary Li wrote: Tested but no effect! Yes but where did you put it? Please post your
Caller ID is normally sent between the second and
3rd ring on an analog trunk. Only way around that is a digital line such
as a PRI.
- Original Message -
From:
Gary
Li
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, November 04, 2005 9:41
Steve,
I see.
Thanks so much for ur quickly answer!
Steve Totaro [EMAIL PROTECTED] 写道：
Caller ID is normally sent between the second and 3rd ring on an analog trunk. Only way around that is a digital line such as a PRI.
- Original Message -
From: Gary Li
To: Asterisk Users Mailing
I have looked
through other postings to the user group for HDLC errors, went through what
worked for other people, and still can not seem to get past this
issue.
For 3 days, I have
been getting HDLC abort(6) errors in *. Prior to Tuesday, the circuits were
clean...I had maybe 10 HDLC
I got Snom to work to flash the MWI and the stutter tone.
A waiting message will change the softkey to VMail. Where do I set that
VMail means dial 8500 ???
bye
Ronald Wiplinger
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Hello all,
On HEAD 10/28/2005 my chan_capi-cm-0.6 is working fine. If I go to
10/29/2005 or newer, something freaks out and I get the following
behavior:
*CLI == ISDN1: Incoming call '19142775896' - '2781980'
-- Executing Set(CAPI/ISDN1/2781980-0, IncomingLine=2781980) in
new stack
--
Intermittently Ill get calls from my only SIP extension to itself via the
Zap/1. I have no clue and have
found nothing online. I have listed my configurations and a
sample of the console messages I see why debugging. Right now it only happens
to the 6000 extension. Any
assistance is
hi all,
I use sipura2000 as sip adapter in our PBX based on asterisk.
It works well exceptcould not get the incoming call's id.
I test several analog telphone, only simens C42 can show the incoming call id.
Any one know what is the reason?
OK I am exhausted.
I can't seem to figure out how to send a caller ID along with a
Outbound call.
Can you believe that I got Vonage to reset my Cisco ATA for $15.00
I then canceled my account!
Well I was with them for over two years, now I am running Asterisk like the
big boys! LOL...
On 11/04/05 21:50 BJ Weschke said the following:
that was built with 3.0 gcc. There are multiple areas in the code
that now use = 3.0 gcc optimizations. It's important that use a
noted. however, i'm still trying to debug a problem which is either with
the freebsd 4.x threading library or
Hi Jesus,
FX is not a variant of R2. It is a completely different signalling
protocol. This means your service provider is using R2 for some of your
channels, and providing all your incoming calls on those channels. It is
use FX signalling for other channels, and you must make your outgoing
On Fri, Nov 04, 2005 at 12:12:08PM +0100, Dave Cotton wrote:
On Tue, 1980-01-01 at 09:11 -0800, Trixter http://www.0xdecafbad.com/
wrote:
A jiffy is a kernel timer, this affects many thing in the kernel.
Linux for as long as I know uses 1000hz. I am really surprised
this failed on fc4.
On Fri, Nov 04, 2005 at 11:43:37AM -0900, Mojo with Horan Company, LLC wrote:
Try putting a line at the very bottom of /etc/rc.d/rc.local like
/sbin/modprobe ztdummy
Which means ztdummy gts loaded only after asterisk is run?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
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