Hello,
-
In several VoIP projects I have very often been asked if IAX is already an
IETF stanadard. I could not anwer this question and I could not find any
IETF draft about this protocol. Can you point me to the right location or do
you know if Digium is going to publish the IAX
Title: Message
Some switches
provide the functionality to try a number till it becomes available. Thus
whenone dials a number and get a busy, one enters a *XX# code and the
switch will call your extension when the called party becomes available. Has
somebody already built this in/for
At 15:30 26-6-2003 -0300, you wrote:
I've tested ATA186 with a cisco827 as the H323 (or SIP) gateway
and I could transmite the fax without problem.
I get erros when sending faxes only when I user asterisk. :~
any tips?
I imagine the Cisco stuff uses T30/T38 amongst
At 17:30 26-6-2003 -0500, you wrote:
toy buy my first cisco 186 but when i read this page
http://www.djernes.org/~shawn/ata186.htmhttp://www.djernes.org/~shawn/ata186.htm
i cant find in my dev page some parameters just like UseSIP
what i need to do to show this parameters
Maybe you have 'Use
I'll have MGCP hardphone that needs to dial pre-defined number as soon as it
goes off-hook. So far I'm lost as to how (if at all) this can be implemented
in Asterisk. Any pointers?
TIA,
Peter
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You can do it with some Agi scripting and call spooling.
Look in app_agi app_qcall.
Matteo
Il ven, 2003-06-27 alle 08:52, Michiel Betel ha scritto:
Some switches provide the functionality to try a number till it
becomes available. Thus when one dials a number and get a busy, one
enters a
I resend this message, as it was not posted on the list first time I send
it
Dan
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 27, 2003 10:13 AM
Subject: Re: [Asterisk-Users] Basic Asterisk questions - personal coments
Why is it that
My bandwith is 64kb/s (ISDN BRI) and so i try to use x-lite which has
many codecs. But i have no audio and i don't see where is the problem.
the calls ring, the connexions are good x-lite - x-lite, x-lite -
phone, there is no drop on the firewall (gateway+firewall+asterisk) and
if i call with an
Please, can anybody help me with this ?
Thanks,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Donnerstag, 26. Juni 2003 19:00
An: Asterisk User
Betreff: [Asterisk-Users] bug in cdr ?
Hi all,
i have a TDM40B
Hi!
Has anyone ever tried to bridge cisco 5300 (talking SIP) and MGCP
endpoint over *?
Seems that there is a bug or something. When * reinvites Cisco for
bridge, Cisco replies with different set of SDP parameters and expect
RTP stream on another port.
regards,
Ekke EInberg
see if your pc has the auto turned on in the main audio control
panel... ;)
i use x-lite and work very well, which codec are you using?
send the trace of sip debug command...
regards,
Angelo
Friday, June 27, 2003, 11:14:18 AM, you wrote:
HT My bandwith is 64kb/s (ISDN BRI) and so i try to
Hi,.
How can I make that Voicemail app to play only my own recorded message
without the default one?
Thanks,
Dan
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Hello,
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial I get a 404 error from Asterisk. Here are my configs and a
dump from sip debug . But if I make a call from a Zap line (see
extension 2382031), it rings the snom phone
sip.conf:
It should do that already.. when you record your busy and unavailable message it
should overwrite the default ones..
Hi,.
How can I make that Voicemail app to play only my own recorded message
without the default one?
Thanks,
Dan
___
Hi,
What's the status of the Gnophone?
www.gnophone.com has not been updated since April 3, 2002...
regards,
--
Jukka Tainio | Kase ry. http://www.kase.fi | tel: 06-8887222
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Hi Dan-
In Steven's defense, he did write back to me outside the mailing list and
told me what he meant. I was using reply in a new thread.
He also answered my technical questions with a brief summary.
But your other comments are interesting - it is difficult for a new user
like me to get
Hi,
Nope.
I have recorded my own busy and unavailable message from the '0' menu of my
voice mailbox.
When someone is redirected to the mailbix, it hears both of them... first my
recorded message, second the default one.
I check that on two separate Asterisk boxes.
I have the latest version from
Hello Everyone -
Well, I think I'm getting closer with the asterisk connection. This is my
setup and I keep getting this error below in ,my /var/log/asterisk/messages
file. I have opened 5060 port on the firewall box.
I would this is Warning which I can ignore! But I see the connetcion
hi there..
I have an asterisk installation with a PRI-E1 card
running EuroISDN, installed on a 1GHz Intel Celeron
box with 256Mbytes RAM.
CPU usage is stuck at 100% all the time, even with
no calls going through. Is this the normal ?
Running top reveals that the CPU allocation is
99.6% to
Don't know.. Are you using voicemail or voicemail2??
Maybe you have found a bug..
Hi,
Nope.
I have recorded my own busy and unavailable message from the '0' menu of my
voice mailbox.
When someone is redirected to the mailbix, it hears both of them... first my
recorded message, second
Dan,
The first question is :
is your voicemail in the default location or have you moved it to another disk?
if you do this you need to update the vm system link in the
/var/spool/asterisk directory eg:
vm - /home/asterisk/voicemail/default/
using ln -s new path vm
also make sure * has the
Voicemail
It is not a bug.. Just do not want to record a separate message like Hi,
I'm unavailable to answer your call...blah blah blah and another one for
Please leave your message after the toneblah blah blah because this
one (the last one) is common for all the mailboxes, so a single voice
Hi Andy,
The first question is :
is your voicemail in the default location or have you moved it to another
disk?
Default location.
if you do this you need to update the vm system link in the
/var/spool/asterisk directory eg:
vm - /home/asterisk/voicemail/default/
using ln -s new path
i do't see what you ask, i go to proprties audio but there is no what
you say.
since i see there is no pb with x-lite on a win 98 but i have problem
with x-lite on win 2000. in sip.conf i have put dtmfmode= info for there
is pdb with GSM and inband
on x-lite codecs are automatically place on GSM
On Fri, 2003-06-27 at 01:59, Peter Zeltins wrote:
I'll have MGCP hardphone that needs to dial pre-defined number as soon as it
goes off-hook. So far I'm lost as to how (if at all) this can be implemented
in Asterisk. Any pointers?
First you must find out if your MGCP phones will automatically
I agree. Besides think of how much more marketable it would be if you
could implement a solution for a client where they would be able to do
some of the more simple changes on their own without incurring an
expense. When they need real work done to their box then they'd call in
the experts. Why
I'm using happily this cheap phones, but I still have a little problem.
Configuring the phone is extremely easy on * and I've a couple of them
perfectly working, except when i try to call some toll-free number (in italy
800xxx ).
If the number called is an IVR system, often with GrandStream
Title: Message
Hello-
Although I'm a very
experienced voice applications designer, C programmer, etc, I'm new to the
Asterisk/Digium environment. I've got an opportunity to use this
software/hardware in an upcoming project which has a near term deadline, and so
I face a steep learning
Anton Yurchenko wrote:
Hello,
The Issue is fixed by setting in snom100 under Settings-SIP- Stack
treat as: to address instead of route.
Than happend becouse somebody has been plaing with the phones without me :)
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial
This can not be done in * directly. Since MGCP phones are dumb,
everything must be done in call agent.
For your functionality the call agent must send ringtone signal to the
phone immediately anfer phone goes of hook.
AFAIK *'s MGCP module can not do this, when a phone goes off hook the
hi,
can some one tell me a good IP phone (not software, but a real
phone :) that work well with asterisk?
how mutch does it cost a good IP phone?
i made a VoIP network for my company, but now we are using a client
for PC phone...
i'd like to buy a IP phone, can someone tell me witch model i
Hi Dan,
[EMAIL PROTECTED] (Dan) writes:
Is your firewall redirecting incoming connections on n.n.n.n:5036 to
the Internal Asterisk instance? If you don't see any messages on the
inside Asterisk box it's unlikely.
More than that... Asterisk is configured as DMZ in the NAT router, so it
On Fri, 2003-06-27 at 02:13, Dan wrote:
Why is it that most users who don't understand threaded email is on
Windows systems.
What do you mean by that Steven?
If the mail list application cannot handle the threads in an intelligent
way, it is our fault??
Its not the mailing list
Yo Dan,
Try adding the s to the arguments you give to VoiceMail2, so, for
example, Voicemail2(sb1000) for the busy-message of ext. 1000.
Note that only Voicemail2 allows the s to be used together with
b and u.
Grtz,
Oliver
On Fri, Jun 27, 2003 at 15:04:44 +0300, Dan wrote:
Try to put
noload = chan_oss.so
in modules.conf
also do you use mpg123 with musiconhold ?
Martin
On Fri, 27 Jun 2003, Dave Alan Caruana wrote:
hi there..
I have an asterisk installation with a PRI-E1 card
running EuroISDN, installed on a 1GHz Intel Celeron
box with 256Mbytes RAM.
CPU
Hello,
If you
have the SIP firmware load in the unit then UseSIIP should be the first
yellow box in the left column. If
you do not have that you may have a MGCP version and need to get a hold of the
SIP firmware and Flash the unit first.
Shawn L.
Djernes
-Original
Hello,
tray http://www.grandstream.com/y-product.htm
BudgeTone-100 costs around $75
I think it's the best price/features voip phone on the market at this
moment :)
Best regards
Lubo
Angelo Sampietro wrote:
hi,
can some one tell me a good IP phone (not software, but a real
phone :) that work
On Friday 27 June 2003 01:40 am, Olaf Menzel wrote:
In several VoIP projects I have very often been asked if IAX is
already an IETF stanadard. I could not anwer this question and I
could not find any IETF draft about this protocol. Can you point me
to the right location or do you know if
I was playing with the agent application to see if I could get it to
work.
Everything works fine, except that Asterisk plays musiconhold while an
agent is logged in and is not taking a call. Is there a way to disable
the music in this situation?
Imagine working tech support where you had to
Hi!
Has anyone ever tried to bridge cisco 5300 (talking SIP) and MGCP
endpoint over *?
Seems that there is a bug or something. When * reinvites Cisco for
bridge, Cisco replies with different set of SDP parameters and
expect RTP stream on another port.
regards,
Ekke EInberg
Have you set
You haven't quite supplied enough data to solve this problem.
Have you successfully used your ATA-186 on Asterisk when they're on
the same network segment (no firewall)? Is your firewall a NAT? It
appears that there is a NAT at both ends of this session. That
probably won't work, if that's
You could create a simple moh class that played a silent mp3 as a very low rate,
or even the occasional beepthen just use setmusiconhold,newclass
hth
Andy
On 27/06/2003 at 13:10 Derek Beaumont wrote:
I was playing with the agent application to see if I could get it to
work.
Everything
On Friday 27 June 2003 01:40 am, Olaf Menzel wrote:
In several VoIP projects I have very often been asked if IAX is
already an IETF stanadard. I could not anwer this question and I
could not find any IETF draft about this protocol. Can you point me
to the right location or do you know if
comment out the music = line in the queues.conf
Derek Beaumont wrote:
I was playing with the agent application to see if I could get it to
work.
Everything works fine, except that Asterisk plays musiconhold while an
agent is logged in and is not taking a call. Is there a way to disable
On Friday 27 June 2003 16:23, Angelo Sampietro wrote:
hi,
can some one tell me a good IP phone (not software, but a real
phone :) that work well with asterisk?
Hi Angelo,
---
I am testing the Snom100 and Snom200 phones. Both working fine under Asterisk.
(http://www.snom.com)
For
No, Asterisk is not a SIP proxy in a true sense of the term.
Almost none of the things you mention below are possible with
Asterisk without additional coding to make them happen. Of course,
if you'd like to submit patches, I'm sure they would be reviewed.
The only two custom SIP headers you
Citeren Shawn L. Djernes [EMAIL PROTECTED]:
Hello,
If you have the SIP firmware load in the unit then UseSIIP should be the
first yellow box in the left column. If you do not have that you may have a
MGCP version and need to get a hold of the SIP firmware and Flash the unit
first.
Oh
Heya all,
I ran some more tests with different kernel-options and my preliminary
conclusion is that the problem goes away when you disable SMP in your
kernel. I even put the Eicon-card, which I suspected was causing the
problem, back into the machine and loaded it's drivers, making calls
through
I'm getting ready to give asterisk another shot
here. Didn't have a lotta luck last time, about 7-8
months back.
I have been scanning the list all this time though,
lurking.
A question that comes up from time to time, that I have
yet to see answered is;
Is anyone actually using * as a primary
Hello Tilghman,
--
IAX is not an IETF standard. However, it is an open protocol (i.e.
not proprietary), as all source is offered under an open source
license. You may also license the IAX code with another license by
contacting Digium, or you may use a clean room
I built * with PROC=i586 and PROC=i386 options in a computer with VIA C3
933MHz processor. When I try to run * I got Illegall Instruction error. Does
somebody know how to solve it?
Please help!
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I'm looking at getting the Dev light applications from digium
and I have 2 Createive Labs voip blasters. The voip blaster supports the
G.723.1 codec. After looking at Gnome meeting it does not talk unless you have a
quicknet card for it. Can I make calls using asterisk and the digium cards to the
Thanks Olivier...
This is the solution.
I never known that the two switches can be used together.
BR,
Dan
- Original Message -
From: The Traveller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 27, 2003 6:48 PM
Subject: Re: [Asterisk-Users] Voicemail issue
Yo Dan,
Try
Hi Simon,
It is solved now It was a problem at the ISP level. some UDP packets
filtered..
Thanks,
Dan
- Original Message -
From: Simon J Mudd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 27, 2003 6:34 PM
Subject: Re: [Asterisk-Users] Asterisk, IAX and NAT issue
For the Snom100 is a IAX Image available at asterisk's ftp site.
Can you tell me more? Is it a patch to enable IAX or replacement firmware?
Many thanks,
Simon
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Dave,
That looks very good. Can't wait to get my hands on it! This is extactly
the type of interface that allows you to retain full control, while
making things easier for the occasional user.
Dylan.
Dave Packham wrote:
OK lets start out with this.
Im not a pro GUI designer ?
Now that
Of course canreinvite=no solves it for outbound calls. But, you don't
want to have 200 RTP streams going through your * server, do you?
I was hoping that someone knows how to get those endpoints to talk each
other directly...
e
John Todd wrote:
Hi!
Has anyone ever tried to bridge cisco 5300
I have several (various brand) sip devices with static IP's.
I understand that asterisk will not accept a registration from these
devices if the host= parameter is not set to 'dynamic' in sip.conf.
I want calls to these extensions to be routable even before the device
registers. I understand
From my testing, it seems that the SIP userid has to match the auth
userID. THis doesn't match my reading of the RFC, and none of the
other SIP servers I've worked with have this behaviour. Is there an
asterisk-specific reason for this, or do we just differ on our
readings of the RFC?
btw:
I'm not sure if immediate mode is enabled but certainly it could be
implemented as it is in zaptel.
Mark
On Fri, 27 Jun 2003, Peter Zeltins wrote:
I'll have MGCP hardphone that needs to dial pre-defined number as soon as it
goes off-hook. So far I'm lost as to how (if at all) this can be
The s flag should cause it to skip the second announcement if memory
serves.
Mark
On Fri, 27 Jun 2003, Dan wrote:
It is about the Please leave your message after the tone.. not about the
busy/unavailable one which are replaced my mine.
How can I disable this message, as it is included in my
What is the module name for the TDM40B - I received my X100P and TDM40B today (thanks
Digium).
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I have a Budgetone and an ATA but none of them
support GSM. I´d like to place call to the PSTN with my X100P viaa WAN
(64kbps). g711 is out of the question. Can * transcode from g723.1
to GSM? How costly is it? I have tried different configurations on
sip.conf and extensions.conf but have
I wrote the one on My Site Djernes.org. If you wish to modify it for
setting up the Device for MGCP then go ahead. I will either post it here or
you can post it and send me a link. I am going to finish the SIP one when I
can dig up some time to write.
Shawn
-Original Message-
From:
The X100P is modprobe wcfxo
The TDM40B is modprobe wcfxs
Andy
*** REPLY SEPARATOR ***
On 27/06/2003 at 16:07 Steven P. Donegan wrote:
What is the module name for the TDM40B - I received my X100P and TDM40B
today (thanks Digium).
___
Greetings! I have made great progress thanks to this
group. My Asterisk seems to be working for the most
part. I am using the following equipment/software:
* HP Vectra VL - Pentium Pro CPU - 256MB RAM
* Redhat Linux 8 - Loaded straight from distro CDs as
Developer Workstation - latest updates
Thank you - modprobe(s) successful.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andy Powell
Sent: Friday, June 27, 2003 4:30 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] modprobe ? for TDM40B
The X100P is modprobe wcfxo
The TDM40B is
Should Asterisk run under it's own user id, or the web server user id,
or root, or what?
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On Fri, 2003-06-27 at 18:06, Mark Street wrote:
I just got a flyer from my buddy on these phones today, totally SIP based,
includes the G.729 speech compression codec.
http://dm.zipphones.com/dm/zip2/index.htm
I don't know how others feel about the callerid display, but since this
one
On Friday 27 June 2003 18:39, Chip G wrote:
* Redhat Linux 8 - Loaded straight from distro CDs as
Developer Workstation - latest updates from RHN
However, the quality of the recording is terrible.
1) Start up in run level 3, not 5 (i.e. /etc/inittab).
2) Turn off frame buffer mode in
Hi John,
I have done this too for Cisco 79x0 with Solar Winds TFTP server, serving
firmware, default configuration and specific configuration for each phone.
Do you have some Ring configuration files to be used with those phones?
Thanks,
Dan
- Original Message -
From: John Laur [EMAIL
Hi,
Check with another software phone to see if the quality is the same.
I have run Asterisk even with GUI installed and started on a Celeron/300MHz
with 128MB of RAM with a couple of connections and it works fine for me.
Best regards,
Dan
- Original Message -
From: Tilghman Lesher
I started using Festival for the first time today and am having a problem
with zombies left behind after every time that it speaks. I'm using
Festival 1.4.3 with today's CVS of Asterisk. Everything seems to work.
The only obvious problem is that a defunct process is left behind every
call to
Netscape has a bug in it that will not allow you to text search into a
editbox. NS is working on fixing that.
Sorry
Dave
[EMAIL PROTECTED] 6/27/2003 1:17:54 PM
OK lets start out with this.
Im not a pro GUI designer
?
Now that thats done. Welcome to OpenConf. At least that what we
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