Ok, no worries then..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Domingo, 20 de Febrero de 2005 02:41 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Soundcard problems?
On Sun,
On Sun, 2005-02-20 at 01:10 -0600, Anton Krall wrote:
> Has anybody had any problems with their soundcards like this:
>
> Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on
> sound device: Resource temporarily unavailable
>
> This shows on the console and I have no clue w
Has anybody had any problems with their soundcards like this:
Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on
sound device: Resource temporarily unavailable
This shows on the console and I have no clue what it is.. voice prompts
sound good
Any clues?
__
On Fri, 2005-02-18 at 16:08 -0500, Giovanni Powell wrote:
> If there is a power failure, which cards other than x100p and
> voicetronix openswitch provide "redundancy".
Once you go beyond analog, you have to have power. Power isn't always
cheap but it isn't hard to come by. I have UPSs at home tha
On Sat, 2005-02-19 at 23:26 -0600, Michael Giagnocavo wrote:
> If you have a TDM card already, buying a T1, channelbank, etc. to add a few
> lines is the stupidest thing I've heard of today.
>
> Have you looked into buying some cheap multiport ATAs? 2 port SIP/IAX2 ATA
> should be around $70-80?
Thank you all for the quick and straightforward response I was looking out for.
Ed, I am an Indian and have parents and friends in India. So that is
the country that I would like to call to.
But let me tell you that although I am al linux geek and can get some
or more work done with it without sw
Well, sure, if you want to spend 8x the amount, yea, it's going to be a much
nicer setup.
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Saturday, February 19, 2005 11:45 PM
To: 'Asterisk Users Mailing List - Non-Commercia
On February 20, 2005 12:26 am, Michael Giagnocavo wrote:
> If you have a TDM card already, buying a T1, channelbank, etc. to add a few
> lines is the stupidest thing I've heard of today.
I disagree wholeheartedly. Recoup your investment on ebay or use it in
another setup. TE110P+Adit600 or TA75
If you have a TDM card already, buying a T1, channelbank, etc. to add a few
lines is the stupidest thing I've heard of today.
Have you looked into buying some cheap multiport ATAs? 2 port SIP/IAX2 ATA
should be around $70-80?
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
Has anyone every setup an external open/close relay, off say a serial
interface, and have an extension trigger the relay?
Why I ask is I have a student accomodation where I am installing an
asterisk box to supply phone services to the tenants, there is already
an intercom system in the main hallw
Guys.
Im testing the default asterisk demo setup that comes after installing, but
I have a problem with dtmf tones... I dial 1000 and listen to the welcome
voice but if I try to enter any keys like 2, 500 etc.. nothing happens...
Why are my dtfm tones not been recognized? what would be the nor
Kurt,
Did you follow the instructions and remove the semi colon in zapata.conf
before the
channel =>1
what happens on the console cli when you make a call.
What type of handsets are you using?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kurt
Fan
Hi, I just joined the list, anyways i am trying to setup an @home box with a
x100p card and so far i can't even get the box to pickup the incoming call
and in the amp management under the section "send calls from PSTN too" page
all the radio buttons are blank and i want to use the digital reception
I have two x100p cards and a TDM20b set up such that the
home phone is one exten, the home office is another, on one of the incoming
lines (the same copper line that has the main home phone number) There is a
distinctive ringing number set up as well. Here’s the crazy part…
any calls comin
If you use the MySQL CDR add-on, you could just query the CDR DB for
the numbers you are tracking. No need to add anything fancy.
On Sat, 19 Feb 2005 21:42:31 +0100, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
>
> Hi all.
>
> I am going to do a simple "voting application" for a radiostatio
Hi Andrew,
Andrew Kohlsmith wrote:
On February 14, 2005 01:18 am, Matt Gibson wrote:
It can receive calls both when receiving power, and when not receiving
power. It can make calls only when not receiving power from the wall. I
tried unplugging it for a good 10-15 minutes to make sure
it was off fo
On Sat, 19 Feb 2005, Alistair Cunningham wrote:
> The issue is with the provider of the PRI. They may or may not accept
> arbitrary callerids on outbound calls. For example, in the UK many
> providers will only accept callerids that are assigned to the trunk on
> their end. You should check wit
Makes you wonder about the future of CISCO doen't it? You are a
potential customer trying every means possible to give them money, and
they are making it difficult to do so. Most thriving businesses
usually make it as convenient as possible for their customers to give
them money.
This re
I could host it on my k-rad 56k sportster USR modem!
-Original Message-
From: Sergey Kuznetsov [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 19, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wiki down?
Or I can host it.
I ha
Race.
Here are thre results of the tests:
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined -
0x0 (nothing)
Seems both can speak ulaw
Really? For five lines I need to buy all that hardware?
Hmm.
Well, I appreciate you taking the time to respond to my question.
Regards,
Jim.
[EMAIL PROTECTED] wrote:
> Digium tech support recommends going with a t1 card and a
> channel bank. This is by far the simplest, cheapest and
> cleane
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Joseph
> Sent: Sunday, 20 February 2005 11:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Anyone used the ACT P104SLD SIP Phone
>
> On Sun, 2005-02
Easily. AGI script + DB.
[EMAIL PROTECTED] wrote:
Hi all.
I am going to do a simple "voting application" for a radiostation.
The idea is to have listeners call in to vote on songs.
What I want to do is to take a phonenumer for each song and present the
result on a simple webpage.
Eg.
To vote on so
Or I can host it.
I have a few colo servers at "1 and 1" with half of terrabyte of monthly
traffic on it. Multiple connections to Tier-1 providers, 12 Gbit total
bandwidth.
I would host it with pleasure.
Nir Simionovich wrote:
Well,
I have no idea where the wiki is hosted, but if the wiki need
On Sun, 2005-02-20 at 08:38 +1000, James Bean wrote:
> Just after some peoples impressions if they have used this phone.
>
> It has 10 function buttons which I am hoping can be individually
> programmed for destination to accept hints from asterisk.
What do you mean by this?
I'm not sure I unders
His Luis,
I have the same setup, the only difference is that I am using Quintum
gateway... is there anybody responded with your concern? I am taking too
much time with this project, in case somebody give you a hint, I wonder if I
could have a copy on it for my reference too.
Your kindness is ve
Thx Race!!! I'll try that and post the results in case somebody else has the
same problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial D
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
This is a very good place to start Race.
This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Grasshopper,
You have your first clue, the live test works.
Do you understand how SIP works? During the INVITE sequence the Asterisk
and the phone trade RTP CODEC information. RTP is the protocol that
actually carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RT
I don't know - they look kinda lame. I mean, why is their SIP server
seemingly better-routed than their IAX server? In my case, their IAX server
is almost 20ms further away than the SIP one -- seems odd to me.
Think I'll stick with Nufone - very well routed, and only ~15ms away. :)
-d
At 06:37 P
I have been using simpletelecom.com for over 2 months now to make
outboudn long distance calls, I didn't have any problems what so ever
with them.
To send callerid this is how I do it:
exten => 81NXXNXX,1,SetCallerID("MY NAME" <1235551234>)
exten => 81NXXNXX,2,Dial(SIP/${SIMPLETELE}/${EXTE
Hi Race..
In this case, the asterisk|home comes preconfigured with some stuff
different than the asterisk tar file.
I check and the phone supports all mentioned codecs, I also made a test by
using the phone and sjphone to do a live test directly, conversation was
successful using gsm, ulaw and il
Good work Edward.
Sometimes it is not you but the machine. Probably a device driver that
was not kosher.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward
Banfa
Sent: Saturday, February 19, 2005 1:00 PM
To: Asterisk Users Mailing List - Non-Com
Okay,
A couple of things could be happening so let's run through a list. Your
questions are a little vague so I shall make my answer also vague.
1. Codec.
Are you allowing for and does the "phone" support the codec that
the sounds are in? (I.e. do you have a G.729 license for your Minke
I installed X100P from DigitNetworks. The system found
the Wildcard X101P and i was able to modprobe zaptel
and wcfxo and ztcfg. then, after i compile asterisk, i
am able to make third party voip calls dialing from
the asterisk cl and using a regular headset. but when
i try to dial out from zap cha
Just after some peoples impressions if they have used this phone.
It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.
Any input would be very much appreciated.
James
___
Asterisk-U
Hi,
I have brought up and asterisk server with a digium T100p. I am
terminating the T1 to a DMS500.
I can make domestic calls fine but when I try to dial international it
gives me an voice error of "77-3 The # you dialed is incorrect please check
the area code and try again". The calls are
I got a message saying that my account was set up, but did not get
programming instructions.
Were the strings I posted correct?
--On Saturday, February 19, 2005 4:54 PM -0500 Brian Dingman
<[EMAIL PROTECTED]> wrote:
You have to wait till you get an email from them saying your account
is setup.
You have to wait till you get an email from them saying your account
is setup. I had the same problem where my DID was setup before my
outgoing account.
On Sat, 19 Feb 2005 13:16:15 -0800, Ed Greenberg <[EMAIL PROTECTED]> wrote:
> I have several DIDs (working well) with LiveVoip and I just signed
I received this email link from Macromedia today
http://www.macromedia.com/newsletters/edge/february2005/index.html?sectionIndex=3&trackingid=AWQX
They are now selling Breeze video conferencing server by the
minute as an asp, this service is powered by the macromedia communications
server
Rob,
From a technical point of view, this is no problem. Asterisk can set
any callerid you like, and the PRI can transport it.
The issue is with the provider of the PRI. They may or may not accept
arbitrary callerids on outbound calls. For example, in the UK many
providers will only accept call
How does he get his offshore relatives into FWD? Nobody said that they have
broadband. Just telephones.
--On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson
<[EMAIL PROTECTED]> wrote:
Or, he could just sign both ends up with FWD and not have to mess
with this *.
>
> I'd like to terminate IAX call on PRI interface. What conditions should be
> met to be able to send arbitrary caller numbers to called party, so he can
> call back to supplied ISDN number (that is different for every IAX user) -
> not through PRI interface, but plain ISDN call !!
That one se
[snip]
> > It's probably made by the same Chinese company that made the cheap GE
> > and Sanyo cordless phones I am using. Whenver your Sipura (I assume)
> > sends the refresh message to turn VMWI off, the phone will display this
> > message for 60 seconds.
> >
> > Change your refresh to somet
I have several DIDs (working well) with LiveVoip and I just signed up for
some outbound minutes. Unfortunately they did not send connection
instructions.
I tried:
exten =>
_1NXXNXX,2,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN}|60|s)
but I get the error
Feb 19 15:14:09 WARNING[21453]: chan_
Or, he could just sign both ends up with FWD and not have to mess
with this *.
> Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
> just calling overseas from Nebraska.
>
> He may also be able to start overseas DIDs that route to his box here in
Wiki is back up.
Between comment SPAM storms, over eager robots ignoring
robots.txt, and mysql issues, it has been an interesting week.
Jim
James H. Thompson[EMAIL PROTECTED]
[EMAIL PROTECTED]
- Original Message -
From:
Roy Sigurd
Karlsbakk
To: Asterisk Users Mail
Hi,
I'd like to terminate IAX call on PRI interface. What conditions should be
met to be able to send arbitrary caller numbers to called party, so he can
call back to supplied ISDN number (that is different for every IAX user) -
not through PRI interface, but plain ISDN call !!
Thanks in adva
Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than
just calling overseas from Nebraska.
He may also be able to start overseas DIDs that route to his box here in
the states.
Rakesh, if this is what you have in mind, let us know and we'll point you
in the right direction.
Hi,
with Asterisk 1.0.1 and bri-stuff-0.1.0-RC4a, and two calls already
established on the ISDN BRI, the third call causes scratches in already
running calls and an answer of an unexisting channel:
Ring on unconfigured channel 0/0 span 2
with Asterisk 1.0.5 and bristuff-0.2.0-RC5, this bug i
Hi,
The link for SuSE 9.x startup script is down too !!!
(http://www.leals.com/~mm/asterisk) !
Can someone have this script ?
Can you send me it or post it somewhere ?
Regards,
Fred
Le Samedi 19 Février 2005 19:13, Roy Sigurd Karlsbakk a écrit :
> hi
>
> is the wiki down again?
>
> roy
>
>
Hi all.
I am going to do a simple "voting application" for a radiostation.
The idea is to have listeners call in to vote on songs.
What I want to do is to take a phonenumer for each song and present the
result on a simple webpage.
Eg.
To vote on song number one, call 555-
To vote on so
Well,
I have no idea where the wiki is hosted, but if the wiki needs to be moved
to a more stable location, our hosting facility in Israel is as stable as
you can get. We have 2 circuit running in, BGP4 and an uplink of 4Mbps. I'm
confident it should be enough, no?
Nir S
- Original Message
I just installed [EMAIL PROTECTED] to see how it works and seems there is a
problem with sounds... I dont hear any announcements or recordings... sounds
are on /var/lib/asterisk/sounds and the logs show this:
-- Created MeetMe conference 1023 for conference '8200'
-- Playing 'conf-onlyper
Trevor Peirce wrote:
exten => 555,1,Dial(Capi/210699&Local/555cell)
exten => 555cell,1,Wait(25)
exten => 555cell,2,Dial(Capi/693555)
I have never done this but I imagine that would solve your problem.
Pretty imaginative! Yes, that should work.
___
[EMAIL PROTECTED] wrote:
I have let's say a reception that is comprised of 2 zap extensions and a mobile
phone to dial using ISDN through Capi.
I want to have a delay before starting dialing the mobile phone so that it
rings only when the call has been unanswered for say 25 seconds.
I tried to us
> > Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk
> > Sendt: 19. februar 2005 19:14
> > Til: Asterisk Users Mailing List - Non-Commercial Discussion
> > Emne: [Asterisk-Users] wiki down?
> >
> > hi
> >
> > is the wiki down again?
> >
> > roy
Yes - no go from here
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk
> Sendt: 19. februar 2005 19:14
> Til: Asterisk Users Mailing List - Non-Commercial Discussion
> Emne: [Asterisk-Users] wiki down?
>
> hi
>
Yes. There are lots of messages in the mailing list archives
regarding this problem, some of them even include things to try. You
didn't see these messages when you searched the mailing list archives?
Alex G Robertson wrote:
Some news.
It is not caused by transmission lines, conectors or a
> is the wiki down again?
It looks like it
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
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Jon,
I've just added the ability to leave feedback on vendors. Having scores
for vendors on voice quality, customer service, etc, is on the wishlist,
though there are quite a few things higher.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
On February 19, 2005 10:56 am, Madhawa wrote:
> Hi List!
> any body use www.simpletelecom.com?
> so anyone here has experience with them? are they a SCAM?
This is -biz material. Spew your "is this a scam" bullshit there.
-A.
___
Asterisk-Users mailing
Hi!
if simpletelecom.com is one of best voip service provider, why they
didn't reply me.
I'm waiting for answer(more than 48hrs):(
anyone know contact phone#?
Thanks
On Sat, 19 Feb 2005 13:15:30 -0500, dean collins <[EMAIL PROTECTED]> wrote:
> Also anyone who says "their bank told them the m
> Message: 1
> Date: Sat, 19 Feb 2005 16:20:31 +0100 (CET)
> From: Remco Barende <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] Dutch VOIP-PSTN provider
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: TEXT/PLAIN; charset=
hi
is the wiki down again?
roy
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
Also anyone who says "their bank told them the money is gone" on a
credit card purchase has to have something wrong with them - just
dispute the charges and problem solved.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif
Madsen - Independent Asterisk
On Sat, 19 Feb 2005 21:56:44 +0600, Madhawa <[EMAIL PROTECTED]> wrote:
> Hi List!
> any body use www.simpletelecom.com?
>
> so anyone here has experience with them? are they a SCAM?
What is it with all the "IS THIS A SCAM?!" emails lately? I'm almost
starting to wonder if its a single entity wit
Hi Race,
Seems like question number five did the trick, I just installed asterisk
on another machine with different hardware specs and asterisk is up and
running, what a beauty!!. Man thank u all for the help, I appreciate it.
:-)
Cheers
Edward "Stack Buffer" Banfa
RADFORMS RESEARCH LABS
JOS, P
Digium tech support recommends going with a t1 card and a
channel bank. This is by far the simplest, cheapest and cleanest
solution that I know of.
Jon.
On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote:
> Folks,
>
> In light of all the troubles people report when running more than o
The feature that would be most useful to me is some sort
of rating feature on things like reliability, quality, etc...
where individual users could rate each provider. I'm not
for sure how best to handle something like this, but my
biggest problem currently is trying to decide which providers
ha
Hi List,
I have a Multitech H323 Gateway MVP400 box with 1
phone on port FXO 1.
I have Asterisk ruuning in Fedora Core 3. Both are in
the same network.
But I can't figure what I have to do in Asterisk to
make that box work. What files I have to configure?
Can anyone help me? I will really appreacia
[EMAIL PROTECTED] wrote:
There is any way to do it or the code has to be modified ?
Not yet, no, but I'm working on this enhancement.
For now you can just use two Dial() statements, one after the other. The
downside to this is that there is a very short moment where no outbound
call will be activ
If you're going to promote your own company, that is fine, but you
should do it in the -biz list.
If you're going to try hard to sound like you are an objective third
party, next time try to remember to set your email address properly.
At least this kind of astroturfing brings some sample sip.c
Hi Race,
OK let me start answering ur questions one by one,
1. I am running RedHat 9.0, i just installed it 3 days ago,
2. I am using gcc version 3.2.2 20030222 (Red Hat Linux 3.2.2-5)
3. I am not using NTP.
4. First i i issued make clean then i deleted all the asterisk
directories /etc/asterisk
I have let's say a reception that is comprised of 2 zap extensions and a mobile
phone to dial using ISDN through Capi.
I want to have a delay before starting dialing the mobile phone so that it
rings only when the call has been unanswered for say 25 seconds.
I tried to use Capi/210699:ww6935
> Come to think of it, why don't soft-phones have web interfaces?
>
> If you have a web accessible phone please tell me about it, off-line
> too, I need to increase my inventory of phone models.
The snom sofphone they just released as a web interface, just like the real one.
In fact, I think it's
Robert Burcham wrote:
I have seen no responses to my earlier post:
http://lists.digium.com/pipermail/asterisk-users/2005-February/089944.html
and my problem persists. Would someone please share
their configs and firmware versions?
I sent you an email (off-list) the other day with configs attached.
Folks,
Here is my problem. I am brand new to Asterisk and infact PBX world,
trying to make IP phone to IP phone talk. Asterisk config files start
form setting up the extentions. I have no clue about the jargon used
in the config files. Is there a place that talks about vr_ready,
agents, extensions
E&M is analogue, not digital...
Not true. You can have E&M signaling on a T1
CAS interface. Likewise T1 CAS interfaces can also be setup for FXS
and FXO signaling. This is just how the robbed bits
communicate. With E&M wink before a remote switch sends a call
to a local
> I'd like to establish way to exchange data between two remote Asterisk
> server. Something like call over IAX and send some structured data.
>
> Any advice ?
I don't know if this could be done thru an IAX call.
What you could do is something like this :
- have a php script on one server that PO
> I am in Nebraska, US.
> I have broadband cable connection at my home. And I have friend and
> family in other country.
>
> Using asterisk and some hardware is it possible for me to call to
> landlines to other countries. whiout the need to go through or take
> any service from say "Vonage" or an
Hi Maya,
I dont have a voicepulse account, do i explicitly need one, from the
comments in the iax.conf file i got the impression that if u dont have a
voicepulse account, u should then comment out some of the
sections/context in iax.conf.Am I right or wrong in trying to run
asterisk with out a voic
Hmmm,
Let's use the single step approach here.
I saw only a few of the prior posts for this problem, so please bear
with me as we start from square one. Also I do not know how the server
is being used and if you can do all these things.
Remember that the hair loss and dental fracturing problems
- Original Message -
From: "Peter Svensson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, February 19, 2005 4:36 AM
Subject: Re: [Asterisk-Users] Which PRI card for EuroISDN ?
> On Fri, 18 Feb 2005, Robert Rozman wrote:
>
> > I wonder
Thanks, Robert. Yes, I _finally_ figured out why I
need multiple extension contexts. I'm now one happy
camper.
Thanks again,
Maya
--- Robert Hajime Lanning
<[EMAIL PROTECTED]> wrote:
>
>
> > Robert, thank you very much for that informative
> > write-up. Of course, I now have more questions.
>
It looks like it's breaking at the iax.conf file. Have
you set up your iax.conf with the registration info
your service provider gave you? It should look
something like this:
register => iaxid:[EMAIL PROTECTED]
So, in my case, I have a line that says
register => myid:[EMAIL PROTECTED]
where myid
Hi,
I'd like to establish way to exchange data between two remote Asterisk
server. Something like call over IAX and send some structured data.
Any advice ?
Regards,
Rob.
___
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Asterisk-Users@lists.digium.com
http://lists.di
Anyone using the UIP200 with *? I am having
difficulty getting the phone to register and *stay*
registered for more than 4 seconds.
The * console shows the UIP200 registering and records
the user agent. The UIP200 displays station name and
time on its LCD, then 4 seconds pass and it shows #1
DIS
Hi List!
any body use www.simpletelecom.com?
I subscribe to www.simpletelecom.com for A-Z termination and paid
US$15.00 and US$70.00 via credit card in two days, but my account has
US$15.00 only. I checked my credit card from the bank and they said me
the payment already paid to merchant.
I've lost
Roger Schreiter wrote:
But when dialing a number, I get:
Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to
create/find channel
After I installed my Digium g729 license, I'm trying to place a call
from my Cisco 7960 and I'm receiving the same error:
Feb 19 09:47:06 NOTICE[2
If you are new to VoIP then by all means get phones that can be
controlled via a web page from the phone.
I will say that SNOM has done a great job with their web interface to
the phones. I would praise others, but all I have seen are the Sipura
and the Cisco ATA, both not real intuitive web pages
Hi,
First i would like to thank the kind people of the list who have
answered my previuos mail, but i am still stuck as asterisk still
crashes upon startup, i have read the install article at
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
and i have search the asterisk archives,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
Michiel,
check out www.budgetphone.nl / www.talkin2ya.
Hi List,
I am a newbie, just came to know about asterisk a few days back while
installing suse 9.2.
I have a question for which I am sorry to say,but I havntread through
all the archives, but AFAIK i didnt get the answer in the archives.
My question is like this.
I am in Nebraska, US.
I have br
Is there a way to make Asterisk wait 3 rings to answer the phone, and
only answer if someone in the house does not?
I just got an idea, perhaps I could set it to answer immediately, and
play a ringing sound, and ring the house phones, then if someone
answers they answer, and if they do not, then i
I had a simular problem receiveing E&M Wink dtmf signals from a MITEL sx200
digital. I had a 10% or so problem with astrisk line timing out before all
the digits were dialed.
I tried all the time outs and even messed with the timing of my MITEL
switch.
It turned out that the problems was solv
Yes It must.
I have tried to make channels not continuous 1, 11, etc and it doesnt work
[]s
Alex G Robertson wrote:
Hi.
Does anybody know if channels in various spans (in TE410, for example)
must be
contiguous, this way.
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2
This is happens because of imperfect HDLC code. I am having the same in
my logs, but quite rare
and on spans which is idle. Therefore this is not an issue with PRIs itself.
I may be wrong, but telco technician checked my PRIs as well, and didn't
find any flaws.
I can tell you more. IT happens wh
Some news.
It is not caused by transmission lines, conectors or anything like that.
The telco tecnician just came here and analyzed the circuit and he got no erros!
He sugested me to loop my PRI port in the balum attached in my asterisk box.
And Surprise...
I got the same errors!
The erro
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