Hi
I'm trying to configure my Asterisk box to provide a simple sample
configuration. It's a mandrake 9.1 box, no cards except a sound card. The
config I am trying to achieve is simply one server, with two SIP clients.
Two issues are cropping up - the first, when I start Asterisk, the sound
Hmm, The host seems to be good, I have no firewall rules in place at the
moment for the local network, and everything is consistantly reachable.
it seems to only happen when a call is hung up/initiated, and when the
program is first started...if that might provide any insight.
Thanks!:)
Chris
Hi Steve,
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
Bluetooth just doesn't do anything very well.
There is something which is done better by 802.11 except the range (see
bellow) and the higher theoretical capacity and bandwidth? The '11Mbps'
value is an ideal one in
Hi all,
we are happy to announce the new test-release of chan_sccp.
The Cisco 7920 support is working now, however, some call handling stuff is
hardcoded for testing.
Also some more in-deep-knowledge of Skinny was archieved, how they
handle calls.
Please test it with your 7940/7960 and 30VIP
Chris,
May be your callerID contains characters that cannot be played by Asterisk.
For example:
if the callerid is 1010, Asterisk will not be able to find and play the
file '' in digits directory.
In your case i guess the caller id starts with
Regards...
Girish
From: Doug Meredith [EMAIL
Are you binding to one particular interface?
I.e., Is bindaddr set to something besides 0.0.0.0?
If so, ifdown your other interfaces, start asterisk, then bring up the other
interfaces.
If not, check your codecs. You need some lines like this in sip.conf:
disallow=all ;
Mike Nash said:
Hi
I'm trying to configure my Asterisk box to provide a simple sample
configuration. It's a mandrake 9.1 box, no cards except a sound card.
The
config I am trying to achieve is simply one server, with two SIP clients.
Two issues are cropping up - the first, when I start
I think what Todd was referring to was to JUST do the signaling proxy on the
Asterisk but not proxying the media. The Asterisk box would ONLY do the
signaling handling between the two endpoints and hang over the media stream
to go directly between the two endpoints. This is a question I was
I do agree with what you have said about Bluetooth, however there are some
distinct advantages. The chipsets are dirt cheap, and they normally use
hardly any power. I do agree that the headsets are a rip, and they could go
way down. I have been involved in the whole networked home which peaked
Dear All,
Just an experience to run by all you experts out there. I have started to
put more VOIP calls into Asterisk, most are pass-through calls and some are
terminating on the Digium card to PSTN. Whenever I get to 10 calls or more,
I would start to get choppy sound. I tried to ping other IP
[EMAIL PROTECTED] wrote:
Are you using the 0.7.1 tar distribution or CVS? I was able to compile
the 0.7.1 Asterisk program/sample config's to get a working system on a PC
with no sound device and no phone interfaces. This system is about as
simple as it can get (except for the 3 fixed disks in
T. Chan wrote:
I think what Todd was referring to was to JUST do the signaling proxy on the
Asterisk but not proxying the media.
This is the definition of a SIP proxy. Asterisk is a PBX that supports SIP, but
not really a SIP proxy. As a PBX, it wants to be in the middle of a call. As an
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/ does not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
Take a look at your memory utilization, you should not be paging/caching any
memory.
Switches are will known not to auto-negotiate properly. All switches, nics,
routers, etc should be manually configured for full-duplex. Make sure each
connection is set appropriately for 1000/100/10 mpbs,
Hi Robert
I took it from CVS, and it compiles nicely (after a little help from the group
here).
My PC has a sound card in it, which works for other applications. I've managed
to get rid of the sound error by turning the 'idle timeout' on the sound
server off, after noticing that playing a
Thanks alot, Ray
Well, looking at cat /proc/meminfo, I am getting like 250M memory cached,
with 512M total RAM, for all the gateways I have, this is quite consistent.
Total Memory usages are always low after reboot and then go up to 450M with
time. I was informed that this is normal for Linux.
Sathya wrote:
Frankie,
Thanks for your response, and BKW too.
I am not 'thrashing' anybody here. This is my experience and I have seen
people posting their experiences (good, bad) with many other voip providers
on this list.
well, I subscribed to NuFone because I've seen your kind of postings
What else are you running on your server? On my server running asterisk and
apache, it has the following:
total:used:free: shared: buffers: cached:
Mem: 261443584 237064192 243793920 55992320 143912960
Swap: 260104192 11231232 248872960
MemTotal: 255316 kB
On Sun, 25 Jan 2004, Christopher Lee wrote:
I've had a closer listen to 400*17 through the handpiece rather than just on
speaker phone, and I get the feeling that the Australian ringing tone must
have been tweaked slightly, perhaps with the introduction of the newer
Ericsson AXE exchanges?
Roy [EMAIL PROTECTED] said:
I want to hear about problems with VOIP vendors. Sweeping them under the
rug isn't going to help. If its a valid problem please post it.
Yup. I understand everyone's reaction because, from what I've heard,
Nufone seems to be very high-quality on the technical side of
Soragan wrote:
Hi,
Anybody has clue of what protocol is using in MSN 6.1? SIP?
AFAIK, MSN doesn't support custom VoIP configuration anymore. 4.7 works,
but not later versions. I guess it's simply a matter of hacking the
registry and configuring the shite manually, but I'm not a Windoze guy
Thanks, Ray
No, I am not running any other programs other than basic OS and Asterisk.
No, I am not using the swapMemory yet, but like you under USED memory, I am
using about 450M or above after the computer has been rebooted for a couple
of days. I am using a Baystack switch and I will try to
That was interesting. Asterisk creates the first cdr entry when the call
file is copied to /var/spool/asterisk/outgoing:
,,271536,callout,,Local/[EMAIL PROTECTED],2,Zap/1-1,Hangup
,,2004-01-25 12:22:54,2004-01-25 12:22:57,2004-01-25
12:22:57,3,0,ANSWERED,DOCUMENTATION
On completion of the
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to
have dtmfmode=rfc2833. However, incoming FWD calls from the dialup
access numbers (such as libretel) need to have dtmfmode=inband. To
solve this problem, I created a second FWD account and configured
sip.conf as follows,
Hi Chris,
Matching the tones isn't hard, if you have the right tools. :-)
Try the free sound editor you will find here -
http://www.speech.kth.se/wavesurfer/ Record some of the tone you are
interested in. Then use wavesurfer to look at it. Choose the n-waveforms
view. Left click over the
Give our telappliant voiptalk service a try. End
user tariff on our web site.
Tan
telappliant.com
voiptalk.org
- Original Message -
From: Daniel
Bichara
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 4:12 PM
Subject: Re: [Asterisk-Users] looking for iax
termination
- Original Message -
From: Daniel Bichara
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 4:12 PM
Subject: Re: [Asterisk-Users] looking for iax termination
Hi,
We have termination based on IAX and SIP at Brazil.
Daniel
Daniel,
I would be interested in the details of
I've read several other responses to your post...
Just an experience to run by all you experts out there. I have started to
put more VOIP calls into Asterisk, most are pass-through calls and some are
terminating on the Digium card to PSTN. Whenever I get to 10 calls or more,
I would start to
T == T Chan [EMAIL PROTECTED] writes:
T Whenever I get to 10 calls or more, I would start to get
T choppy sound. I tried to ping other IP addresses from the Asterisk
T and noticed a big packet loss in the vincinity of 7% to 10%, ...
What does your /proc/interrupts look like?
Which kernel are
Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to load
config iax1.conf
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Sunday 25 January 2004 11:19, Philipp von Klitzing wrote:
Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to
load config iax1.conf
As a matter of chan_iax slowly moving towards the deprecated pile, to be
replaced everywhere with chan_iax2, chan_iax now looks for a config
Awesome, that worked! Thanks :)
Chris
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 1:31 AM
Subject: Re: [Asterisk-Users] SayDigits
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
Has anyone had this problem:
Tilghman Lesher wrote:
On Sunday 25 January 2004 11:19, Philipp von Klitzing wrote:
Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to
load config iax1.conf
As a matter of chan_iax slowly moving towards the deprecated pile, to be
replaced everywhere with chan_iax2, chan_iax
Frankie Gravato wrote:
I've been beating my head for 5 hours to figure out why my asterisk
server or sipura isn't passing my voice over to the caller. It seems i
can hear the caller but they can't hear me it seems either the
asterisk or the sipura isn't passing this information.
Here's
I have a similar set up, I don't have a separate sip phone, but I have the
same exact problem with the line 1. I don't know if my config files aren't
right, but I can't transfer between exts yet, but my issues is with line one
on an incoming call from an X100P as well.
-Original
I have had several installations where I was unable in any configuration to
make the FVS318 work with VOIP traffic. I don't belive it is related to any
paticular Phones or VOIP GW have see same problems with even Cisco 7960's
Has anyone opened a ticket with Netgear on this issue?
- Original
same here, when i recive an incoming call from x100p to line 1 on
sipura, i can hear them but people can't hear me im using 1.0.24 on my
firmware
Miguel
On Sun, 2004-01-25 at 20:54, Chris Higgins wrote:
Frankie Gravato wrote:
I've been beating my head for 5 hours to figure out why my
I am trying to find an example of how to set up my FXS Station Card in my
Asterisk.
I have (1) XP100P
I have (1) tdm20B (2 Port FXS)
Could someone tell me if this is correct?
/etc/zaptel.conf
fxsks=1
fxoks=2
fxoks=3
loadzone=us
defaultzone=us
/etc/asterisk/zapata.conf
[channels]
;
This is my working config for x100p tdm400 so if you change the channel
entries from 2-5 to 2-3 you should be good to go.
/etc/rc.d/rc.local
modprobe wcfxo
modprobe wcfxs
/usr/sbin/asterisk
/etc/zaptel.conf
fxsks=1
fxols=2-5
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf
[channels]
I added the music on hold feature. I answer on line 1, flash for a sec and
come back and transmission both way is fine, just can't answer initially.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Miguel Cavazos
Sent: Sunday, January 25, 2004
Steven E. Frazier wrote:
I have a similar set up, I don't have a separate sip phone, but I have the
same exact problem with the line 1. I don't know if my config files aren't
right, but I can't transfer between exts yet, but my issues is with line one
on an incoming call from an X100P as well.
; FXS Port 1
context=local
signalling=fxs_ls
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
;
;FXS Port 2
context=local
signalling=fxs_ls
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
Change the signalling here to fxo_ls. Its gotta match what's in zaptel.com
On Sun, 25 Jan 2004, Martin wrote:
Hello.
I'm not sure what this problem is. Probably one of my libraries.
Any clues ??
---cd ../asterisk then make clean ; make install
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude
On Sun, 25 Jan 2004, Steve Underwood wrote:
Actually, nothing would use a 17Hz tone - it doesn't pass through a
300-3400Hz channel very well :-)
It's not a 17Hz tone. Australian (and others) tones are single-frequency
tones that are amplitude-modulated at a second, much lower, frequency.
I have tried to get my TDM400P card to automatically dial a number or run an
application when I pick up the phone without much luck. After reviewing the
email archives, config files and source to chan_zap.c it appeared that
all I had
to do was set immediate=yes in the zapata.conf file and have a
It would probably help if you used a packet sniffer (eg, ethereal) to look
at the traffic, or at least provide the list with a useful clue other then
it doesn't work.
same here, when i recive an incoming call from x100p to line 1 on
sipura, i can hear them but people
I have Asterisk running with a combination of SIP
and H323 clients. I am using the OH323 module instead of the H323
one.
When the SIP clients ring each other, they can hear
a ringing noise in the ear peice to let them know that the other parties phone
is ringing. However, when the H323
Sathya wrote:
Folks,
I've ordered a new account from Nufone last month. Transferred money to
Nufone through their paypal account. I had communication with Nufone sales
up until two weeks back. Since then there were no replies to my emails.
How quickly you forget that I assigned you a
Cees de Groot wrote:
The fix is not to tell people to shut up, the fix is
that Nufone spends ~2 hours on the website and integrates a decent
trouble-ticketing system (another 2 hours, and if that's too much they
can outsource it with me ;-)).
Who says we don't run a trouble ticketing system?
Do you have bison and/or yacc installed?
Greg Boehnlein wrote:
On Sun, 25 Jan 2004, Martin wrote:
Hello.
I'm not sure what this problem is. Probably one of my libraries.
Any clues ??
---cd ../asterisk then make clean ; make install
gcc -pipe -Wall -Wstrict-prototypes
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Vic Cross
Sent: Monday, 26 January 2004 9:31 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Indications
On Sun, 25 Jan 2004, Steve Underwood wrote:
Actually,
On Sunday 25 January 2004 18:17, Clif Jones wrote:
I have tried to get my TDM400P card to automatically dial a number or
run an application when I pick up the phone without much luck. After
reviewing the email archives, config files and source to chan_zap.c
it appeared that all I had
to do
On Sunday 25 January 2004 12:07, Olle E. Johansson wrote:
Tilghman Lesher wrote:
On Sunday 25 January 2004 11:19, Philipp von Klitzing wrote:
Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to
load config iax1.conf
As a matter of chan_iax slowly moving towards the
Christopher Lee wrote:
On Sun, 25 Jan 2004, Steve Underwood wrote:
Actually, nothing would use a 17Hz tone - it doesn't pass through a
300-3400Hz channel very well :-)
It's not a 17Hz tone. Australian (and others) tones are single-frequency
tones that are amplitude-modulated at a
I think discussions about which VOIP service providers are best
or not best is a reasonable topic for an asterisk user's
list. After all, selecting a provider is a big deal
Rants about some company just make the rant-er look bad and
should be avoided.
Back to NuFone. I've just started testing
On Sun, 25 Jan 2004, Eric Wieling wrote:
Do you have bison and/or yacc installed?
Yes, but that isn't the problem.
I used strace on the bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0)
= 0xbf31e000
_llseek(4, 0,
From: Christopher Lee [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Indications
Date: Sun, 25 Jan 2004 15:49:42 +1000
Organization: Data Chaos
Reply-To: [EMAIL PROTECTED]
Hi Steve,
Interesting... I'm not sure! My copy of the original indications.conf had
400+17,
funny... I got an immediate response, and within 1 hour had my account
activated. and this was today.
-Original Message-
From: Chris Albertson [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 10:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Has Nufone gone
Can anyone recommend a provider to work with that works with managed
services companies? LNP is key for us. From my reading it looks like
NuFone is for personal accounts.
Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sean Cheesman
Sent:
I tried a couple times to talk to them about service. How much it costs,
how it works, etc. Just common stuff you might find on a website. I left a
message and nobody returned my call; I went with voicepulse instead.
John
- Original Message -
From: Sean Cheesman [EMAIL PROTECTED]
To:
John Baker wrote:
I tried a couple times to talk to them about service. How much it costs,
how it works, etc. Just common stuff you might find on a website. I left a
message and nobody returned my call; I went with voicepulse instead.
No messages were ever received from you, thus we never
Which is the best way to contact Nuphone.?
On Sun, 25 Jan 2004, Jeremy McNamara wrote:
John Baker wrote:
I tried a couple times to talk to them about service. How much it costs,
how it works, etc. Just common stuff you might find on a website. I left a
message and nobody returned my
Ryan Finnesey wrote:
Can anyone recommend a provider to work with that works with managed
services companies? LNP is key for us. From my reading it looks like NuFone is for personal accounts.
We have many happy wholesale and retail customers that simply do not
care to opine in this thread. I
I don't use Nufone. But just seeing Jeremy's reply make me want to say
something. As an outsider, if the attitude is No messages were ever
received from you, thus we never called you back. or How quickly you
forget that It makes a customer wonder the attitude the staff at
Nufone has. Again
Here here!!!
--- David Liu [EMAIL PROTECTED] wrote:
I don't use Nufone. But just seeing Jeremy's reply
make me want to say
something. As an outsider, if the attitude is No
messages were ever
received from you, thus we never called you back.
or How quickly you
forget that It makes a
Folks,
Thanks for brick bats as well as back rubs :)
I started this thread, so I have to close it.
I deal with many customer in my work and they are never wrong. I always
learn from them. If a customer sign-up with me and if I can't offer the
service immediately, I am loosing my opportunity, so
David Liu wrote:
I don't use Nufone. But just seeing Jeremy's reply make me want to say
something. As an outsider, if the attitude is No messages were ever
received from you, thus we never called you back. or How quickly you
forget that It makes a customer wonder the attitude the staff at
Maybe we can setup a time to talk sometime this coming week and see if
we can come to some type of wholesale agreement.
Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jeremy McNamara
Sent: Monday, January 26, 2004 12:05 AM
To: [EMAIL
Dear All,
I will send you a couple more emails, the story is something like for some
reasons, Nufone website was not working for a bit of time or something, and
this and that. So, there were some customers / potential customers who
posted in the asterisk discussion forum titled 'Has Nufone gone
You're right, Jeremy. I made up the whole thing. I went out of my way to
concoct a story about how I wanted to do business with you, but was unable
to figure out how on your website, so I called and left a message and didn't
get a return. Yeah, whatever.
Check your logs. About 3-5 weeks ago.
John Baker wrote:
You're right, Jeremy. I made up the whole thing. I went out of my way to
concoct a story about how I wanted to do business with you, but was unable
to figure out how on your website, so I called and left a message and didn't
get a return. Yeah, whatever.
You could have
71 matches
Mail list logo
