Bill Reid wrote:
For the last few days I can not resolve voip-info.org from many DNS
servers. It does resolve with some DNS servers but I suspect it may be
related more to caching.
I've alerted James of the problems. I haven't seen them myself, so its hard
for me to track.
The wiki has become
It seems to be a Grandstream specific thing. Prolly specific to certian
GS firmware revs
On Sat, 2003-12-13 at 15:18, Brian West wrote:
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
On Sat, 13 Dec 2003, Philipp von Klitzing wrote:
Incorrect
Hello
I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can
call from IP 7905G to SNOM 200 but not the other way round. Instead I get FORBIDDEN
Message on SNOM 200 LCD when ever I try to call IP7905 phone and asterisk generate
following messages..
Please note
Hi!
I don't get why people always say dtmfmode=info mine works fine with
rfc2833.
bkw
Dunno. I tried rfc2833 first, and had exactly the same problem as
described below with voicemail (but only there). Info then worked just
fine (as obviously also confirmed by this user here).
Is there
Just to repeat - [EMAIL PROTECTED]
Please reply if you wish.
- Original Message -
From: Stephen Wingfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 13, 2003 12:26 PM
Subject: Re: [Asterisk-Users] * Party in Paris
SATURDAY 20th
I have had far fewer emails than
If you want a mirror of you site, I can
give some space (how much is needed) , plus
mysql,php, blah blah blah
For free, of course.
Matteo.
Il dom, 2003-12-14 alle 10:00, Olle E. Johansson ha scritto:
Bill Reid wrote:
For the last few days I can not resolve voip-info.org from many DNS
For the last few days I can not resolve voip-info.org from many DNS
servers. It does resolve with some DNS servers but I suspect it may be
related more to caching.
I've alerted James of the problems. I haven't seen them myself, so its
hard
for me to track.
The wiki has become a too
Thank you for all the offers. I think the list would prefer if you mailed me off-list :-)
A gentle remark, no attack. You are really most kind to offer your assistance.
And if any other users have any problems and can track them down, I would appreciate
you mailing me so we can make the Wiki
All,
Please excuse my last post. I meant to send to Olle directly (off-list),
and was operating on not enough sleep.
Of course, I picked the best message of all to send back to the list.
Thanks,
Pat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Hi,
I'm getting this error and don't know how to fix it.
IAX2 seems to work though...
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring
Hi,
could anyone please provide a working sample of how to configure asterisk to
connect to fwd?
I've
tried the one at www.loligo.com and it
doesn't work. Not even when calling to 5.
Can
you advise on how to debug sip (or trace and view sip packets) from the
asterisk server to fwd
I'm getting this error and don't know how to fix it.
IAX2 seems to work though...
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
WARNING[1074439936]: File chan_iax2.c, Line 5465 (set_config): Ignoring
If U want I can send U my settings. My FWD is
working fine. Let me know...
Chris HARIGA
- Original Message -
From:
Shoval
Tomer
To: [EMAIL PROTECTED]
Sent: Sunday, December 14, 2003 10:31
AM
Subject: [Asterisk-Users] Asterisk and
fwd
Hi, could
Hi all,
I'm trying to get a handoff between me and a carrier going using Asterisk.
I need to handoff using CAMA signaling. On a Cisco, you can see the
configuration types that I'm referring to on this site as an example:
Hi all,
I am trying to setup a ZAP interface to do MF signaling for a handoff to a
911 tandem. The signaling I need to perform on the T1 is this:
9-1-1 Tandem: Wink
CLEC end office: KP (Keypulse) NPA ST (Start)
9-1-1 Tandem: Wink
CLEC end office: KP I (Info Digit) NXX ST
As I'm not as
-
Original Message - From: "Alastair Maw" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 12, 2003 4:58 PM Subject: Re:
[Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
On 12/12/03 13:56, Dan wrote: This is because
the fax is transmitted using the
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower
Sent: Sunday, December 14, 2003 12:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
The question asked here, why on earth you want to push fax data
over a VoIP link
Shoval Tomer wrote:
Hi, could anyone please provide a working sample of how to configure
asterisk to connect to fwd?
I've tried the one at www.loligo.com http://www.loligo.com/ and it
doesn't work. Not even when calling to 5.
Check the Asterisk FAQ at http://www.voip-info.org
Can you
At 12:29 AM -0800 12/13/03, SW wrote:
Hi folks,
To provide MWI, * will send out a sip:notify message to the UA.
The originator of this message is asterisk, as shown below;
NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21
From: asterisk
http://www.loligo.com/asterisk/misc/WiSIP/
Works decently enough. Still some software bugs to work out.
JT
At 10:10 AM -0600 12/12/03, Michael Graves wrote:
Anyone here have any good/bad things to say about first hand experience
with the new Wifi SIP phones? I am considering one for my office
thats an old review jeff pulver said firmware was better now, is it
worth 250USD? im thinking of one
does it autodetect ssid? works with low signal? sound quality?
Miguel
On Sun, 2003-12-14 at 18:34, John Todd wrote:
http://www.loligo.com/asterisk/misc/WiSIP/
Works decently enough. Still
Looking at the code in chan_zap.c, I only see options for feature group B and
feature group D MF. The 2 stage MF signalling you are asking for isn't
implmented in the latest asterisk source code.
I would suggest you post a feature request detailing your 2 stage dialing
requirement to
Hi guys,
I think I posted on this issue before, but didn't get a response. I've
still not been able to resolve the issue.
I've got a small installation of Asterisk running one 4 port FXS Digium
card and 1 FXO Digium card. I'm having difficulty routing modem call
through one of the
Hi,
How can I do to register an Asterisk server using just IAX2?
If I have a line like the following in iax.conf
register = user:[EMAIL PROTECTED]
the server tries to register with both IAX and IAX2.
if the line is :
register = user:[EMAIL PROTECTED]:4569
then the same, tries both on the same
Just noload chan_iax right?
Mark
On Sun, 14 Dec 2003, Dan wrote:
Hi,
How can I do to register an Asterisk server using just IAX2?
If I have a line like the following in iax.conf
register = user:[EMAIL PROTECTED]
the server tries to register with both IAX and IAX2.
if the line is :
Hi,
From: Mark Spencer [EMAIL PROTECTED]
Just noload chan_iax right?
Mark
Yes, but... I still need IAX(1) till the problem with DIAX will be solved.
When a call is placed through Asterisk using IAX2 and DIAX, the call is
automatically droped apparently without any reason, after about 60'. It
hi there. i've got a question about outbound dialing.
here's my scenario:
1. i build a list of phone numbers from a database
2. when a call comes in, i begin dialing from the list
3. when an outbound call is answered, i connect the caller to that line.
so far, i'm able to do this with an agi
Hi
I have the following configuration at home one ZAPTEL interface connecting
to an FXO card and two SIP UAs connecting to asterisk locally. I have
configured extensions.conf such that dialing 9 on the SIP phones allows me to
dial an outbound number via the FXO interface . Works fine.
Hi,
Citeren Mark Spencer [EMAIL PROTECTED]:
Just noload chan_iax right?
Mark
Hmm, I understand there is little or no desire to really work on this type of
issue, but noload chan_iax is slightly too rigorous for me. Consider this
scenario:
I prefer using IAX2 with every partner I can talk
Thanks, John.
My requirement here is little different. I am using * as a Voice Mail server
for Vocal ((in addition It does codec conversion and routing to PSTN/SIP and
PSTN/H323). Thing is Vocal doesn't seems to like the Notify message coming
from user asterisk. If I can modify this I will have *
When I run the asterisk System command my asterisk crashes. When I
monitor the console this is the error I get. Any suggestions?
exten = 1,1,System(ls)
this is the error I get:
[EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
Broken pipe
Ouch ... error while writing
Please do.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Chris HARIGA
Sent: Sunday, December 14, 2003
8:41 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Asterisk and fwd
If
U want I can send U my settings. My FWD is working fine. Let me know...
Am I
assuming that a GS set to early dial to * dosn't work. Or am I missing
something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's
not/supported or GS bug, only asking because it's assumptions that alwas get me
:-)
GS
firmware 1.0.4.26
Thanx
in advance
John
it doesnt work here, same firmware 4.26 tryed it with 4.18 also and it
doesnt work, i press any number and it gets screw up
i will try it with the handytone ata286 and see if it works, anyway its
the same firmware but its worth to try out
Miguel
On Sun, 2003-12-14 at 21:13, John Breeden wrote:
On Sun, 2003-12-14 at 12:18, [EMAIL PROTECTED] wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ProvoCityPower
The question asked here, why on earth you want to push fax data
over a VoIP link at all. Fax compression isn't very efficient. may
speak volumes about the
Hi All,
i received my X100P and Grandstream phone last
week. i started configuring my * and with the help of ur mailing lists i was
able to configure it. (when ever i got struck i searched this list and found my
answer. thanks a lot and this list is awesome). i still hv a small problem and
- Original Message -
From: Balaji NJL
To: [EMAIL PROTECTED]
Sent: Monday, December 15, 2003 8:47 AM
Subject: [Asterisk-Users] unable to configure my Grandstream phone
snip
Attempting native bridge of SIP/2003-b895 and SIP/2000-53e2
WARNING[5126]: File chan_sip.c, Line 1954
(found on dslr voip forum)
http://www.freephoneproject.com/nexthop/
I think this is pretty cool. not too sure how stable/durable though, at
least not until there are a lot more nodes providing service. also
unsure of the legality vs. your average telco's consumer tos, but it's
definitely not
Dear All,
I am a new user of Asterisk
interested in setting up a VOIP network based on Asterisk. I have deployed a few
Asterisk servers running on T400P and have started a few weeks ago to run some
LIVE traffic on one of the servers. Most of my current traffic is via H323 to
and from other
Title: Message
Has anyone
succesfully integrated * with a cisco voice gateway ?
This is almost certainly not an Asterisk-specific posting, but due to
my inability to find a VoIP-focused Cisco list, I'll post here in the
hopes of finding a more diverse user community.
I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and
have been experiencing situations
Hi Paul,
thanks for the quick response. i tried the following
configuration /
combination still no luck
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
;context = default
disallow=all
allow=g729
allow=gsm
allow=ulaw
when i tried g711 i am getting an error in * that
codec not
Sip phones generate their own dialtone. The ignore pat option is meaningless
with regard to SIP phones. I would check the Qrandstream's dialplan and see if
you can program it to ignore the dialtone after a '9' is pressed. I had to do
something similar for my Sipura SPA-2000.
Steve.
On
ProvoCityPower wrote:
The question asked here, why on earth you want to push fax data over
a VoIP link at
all. Fax compression isn't very efficient. may speak volumes about
the future role of VOIP. My plans are to role out a VOIP connection to
thousands of Customers. Many have legacy fax
i hv also added the alaw
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
;context = default
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw
now i am able to call from my MSN - * -GS but the
other way is not
working. i am getting lot of noise when i try to place
So do I have this right? You have a 7960 hooked up to your network. You
have a PC plugged into the second Ethernet port of the phone. This 7960
freezes.
If you move the phone somewhere else in the office, without the PC, it
doesn't freeze? Does it freeze when it's in the same location, but the
How is Vonage doing it?
http://www.vonage.com/features_fax.php
John Breeden
Hawaii
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Underwood
Sent: Sunday, December 14, 2003 2:24 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FAX, IAX
Hello John,
I have a similar story with one of mine..
If i use a cell phone near it, it reboots :)
The other ones are fine.. I think it's a hardware problem, but it only
reveals itself with the cell phone radiation :
Sunday, December 14, 2003, 10:59:14 PM, you wrote:
JT This is almost
When I run the asterisk System command my asterisk crashes. When I
monitor the console this is the error I get. Any suggestions?
exten = 1,1,System(ls)
this is the error I get:
[EMAIL PROTECTED] asterisk]# Ouch ... error while writing audio data: :
Broken pipe
Ouch ... error while writing
Using latest cvs? What distro? because it doesn't happen to mine.
bkw
On Sun, 14 Dec 2003, Kevin wrote:
When I run the asterisk System command my asterisk crashes. When I
monitor the console this is the error I get. Any suggestions?
exten = 1,1,System(ls)
this is the error I get:
Thanks for the feedback on this. I just downloaded the latest CVS and
I'm not sure why it's doing this.
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 14, 2003 8:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Crash System
Did DVD
players have to accommodate VHS tapes? Did VHS players have toaccept
beta? Why does VoIP have to deal with an accent protocol that can't
handlelossy audio, nor irregular delays?Also why should we
be soo wasteful when fax machines need a 80K codec toget the data across
IP, and the
Hi Jeff,
I live in Provo and I think I understand the application you're
referring to. Some folks in my neighborhood have been getting to be the
beta testers for these cool new fiber links that the city is supposed to
be laying out. If I only lived a few blocks over, I would be able to
get
yes. Cisco 2612 Router with 2
x FXO's and 2 x FXS's. Works well using H323, and gnugk.
Steve.
Bruce Hedreen [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
15/12/2003 09:57 AM
Please respond to asterisk-users
To:
[EMAIL PROTECTED]
cc:
Subject:
[Asterisk-Users]
It's
just my lowly opinion but I too must agree when it comes to the consumer/soho (1
to 3 line) markets.
CAUTION!!, DANGER!! Marketing Hat
On!!
Vonage, the most "visible" marketer of a voip consumer
product must also agree. Vontage offers anip "fax line".
usingcisco's ata. Vontagemust
It's just my lowly opinion but I too must agree when it comes to the
consumer/soho (1 to 3 line) markets.
CAUTION!!, DANGER!! Marketing Hat On!!
Vonage, the most visible marketer of a voip consumer product must also
agree. Vontage offers an ip fax line. using cisco's ata. Vontage must
see
If Asterisk is configured as a simple answering machine replacement
with the X100P connected to PSTN line. No FXS ports in the
Asterisk machine. Standard phones are connect in parallel with
the X100P like you would a regular answering machine.
Can Asterisk detect that a phone has been picked up
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Sharp
Sent: Sunday, December 14, 2003 6:01 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
It's just my lowly opinion but I too must agree when
I noticed that on earlier versions of the firmware I was able to crash these
phones with a flood ping (ping -f phone-ip). I send this notice to Cisco
but never followed up on it. How are you powering this phone, POE or wall
cube?
-Bill
-Original Message-
From: [EMAIL PROTECTED]
I was playing around with conferencing tonight. I was able to place a
bunch of SIP phones and a couple of my Zap FXS phones into a conference.
So I thought, Let's see what it's like when people come in from outside.
So I called a friend and had him call in on one of my Zap channels,
WHICH IS
On Mon, 15 Dec 2003, Brian Capouch wrote:
Anyone know of a way of doing this when the scumbag ILEC won't give you
supervision?
Probably not much. Try turning on callprogress and/or busydetect - it
MIGHT help. But the only way to do this right is with supervision on an
analog trunk or with
Hi Steve,
- Original Message -
?From: Steve Underwood [EMAIL PROTECTED]
...
. However, I think this has nothing to do with the
original poster's intent.
You're right.
Let's go back to my original problem.
There is any chance to make RxFax work with any type of fax machine?
It works for
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