Hi David,
I've got probably the same/a similarproblem. Do you
add the phones to the queue (AgentLogin/AddQueueMember)?
If there are entries like: " Spawn extension
(macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have
the same problem like me.
I suspect that
On Thu, 2005-09-15 at 16:10 -0400, David Sampson wrote:
I’ve reduced my problem down to this:
[EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make
Are you trying to use make from the apps directory?
You have to run make from the main asterisk source directory.
Look at the
Here is conf example
[51]
type=friend
username=Test
secret=testpassword
host=dynamic
canreinvite=no
context=sip
disallow=all
allow=alaw
dtmfmode=rfc2833
And you have to make phone conf. Like this
Username = 51
Password = testpassword
Phone number 51
[EMAIL PROTECTED]
Iyi
Hello,
I have see a post in the list saying that the 'daemon' command should be
remove from the asterisk startup script in /etc/rc.d/init.d/ for FC2 in
order for asterisk to auto restart when crash.
I wonder if this should be done on FC3 as well, because my asterisk did not
restart when crash.
Hi;
I have a Wildcard TE110P, i'm locking for a Motherboard to use it, i
know that in the web site of Digium are 3 Motherboard but it's to
expensive.
Do you tested other Motherboard.
Thanks
--
coordialement
Karim AMER
___
--Bandwidth and Colocation
Hello,
I have [EMAIL PROTECTED] 1.5 installed and all is working fine for
incoming calls and sometimes outgoing calls. Installed in the box is a digium
TDM04B (4xFXO Ports)
setup as ZAP1 to
ZAP4. I have incoming calls coming in on lines 1-4 in that order and
outgoing calls prefering ZAP4
canreinvite=no
- Original Message -
From: Kevin Bockman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 15, 2005 9:48 PM
Subject: Re: [Asterisk-Users] Call recording between SIP phones
Lakmal
I am looking for a simple way to forward calls unconditionally with Asterisk.
We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary,when she is out,she needs to be able to forward unconditionally to her mobile or collegue.
I am trying to keep
Hi,
I am using asterisk 1.0.9 on FC3 box. mpg123 is not working it starts
and stops immediately..
printing -
-- Started music on hold, class 'default', on channel 'Zap/1-1'
-- Stopped music on hold on Zap/1-1
Unknown option: --mono
There is no such option defined in my musiconhold.conf file
OK, I wonder if I have something wrong -- I have the *1 in my
features.conf for the one touch record -- now I called a number, and
when the call was answered flashed the hook and pressed *1 and went
back tothe call, but nothing happpened. I am using CVS from 8/26 --
is this too old or am I doing
hi,
can anyone please tell me under which circumstances asterisk creates
Local/Zombie channels and how to suppress this? It only seems to happen when a
user calls himself, but I can't reproduce this in our testsystem and it only
happens occasionally. All we do in the extensions.conf is send
In extensions.conf I have changed the queue command from:
exten = 1,1,Queue(itsupport|n|||50)
to:
exten = 1,1,Queue(itsupport|tT|||50)
Now it works as it should.
Regards
Wolfgang
Am Donnerstag, 15. September 2005 08:21 schrieb Wolfgang Lumpp:
Am Mittwoch, 14. September 2005 19:12 schrieb
Are there drivers available for Xten Softphones?
Do they work with X-lite/Eyebeam Softphones?
On 9/16/05, Bill McCready (PCPhoneline.com) [EMAIL PROTECTED] wrote:
Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with
c waddy schrieb:
I am looking for a simple way to forward calls unconditionally with
Asterisk.
We are running an Asterisk system with 10 extensions using SIP. One of
our users leaves the office regulary, when she is out, she needs to be
able to forward unconditionally to her mobile or
Hi Stefan,
Has chan_sccp matured a lot? I remember (maybe a year ago or so) that I
had a lot of problems with chan_sccp and chan_skinny (one thing is that
I remember with chan_sccp is that the VM button didn't work and trying
to answer multiple incoming calls tended to make the phone go into
All, I've asked this once a long time ago and got a vague response, any
suggestions? I'm wanting to convert for example a 7 digit extension
(whether it be via dialplan or agi) to e.164. This is for the sake of
getting everything outbound into e164 format. The issue I see you will
need to append
On Thursday 15 September 2005 23:54, Jason Kim wrote:
I tried both 1.0.9 and 1.2beta.
I couldn't see any interrupt from /proc/interrupt.
My email server has no spam filter.
rmmod wct4xxp zaptel (ignore any errors)
dmesg -c /dev/null
modprobe wct4xxp
dmesg -c /tmp/dmesg.wct4xxp
what is the
Thank you for your response.
Can i make the same configuration with I4L (how?) ?
The EICON DIVA PCI 2.02 (as i know) is not usable with CAPI, but only
with I4L.
Am i wrong ?
Massimo Frisoni
Emanuele Pucciarelli wrote:
Massimo Frisoni ha scritto:
I have an EICON DIVA PCI 2.02 with I4L.
hello
I'm working with realtime and oracle.
I'm using two tables in oracle (sip_conf and voicemail_conf)
My extensions.conf looks like this:
[datab]
exten = _3XXX,1,Dial(SIP/${EXTEN})
exten = _3XXX,2,Voicemail(u${EXTEN})
exten = _3XXX,3,Hangup
exten = _3XXX,104,Voicemail(b${EXTEN})
exten =
Asterisk don't running, because show this message
Sep 16 15:28:42 WARNING[1075709024]: cli.c:702 ast_cli_register: Command
'showparkedcalls' already registered (or something close enough)
== Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
If I understand correctly you are supposed to patch the Makefile in the apps
directory and then run the main Makefile. I've tried both ways - the patch
failed on the main Makefile. Should I try to make that work?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
On Fri, 16 Sep 2005, Massimo Frisoni wrote:
Thank you for your response.
Can i make the same configuration with I4L (how?) ?
The EICON DIVA PCI 2.02 (as i know) is not usable with CAPI, but only with
I4L.
Am i wrong ?
I don't know the current status, but mISDN and its CAPI should support
Hi Derek
Derek Conniffe schrieb:
Hi Stefan,
Has chan_sccp matured a lot? I remember (maybe a year ago or so) that I
had a lot of problems with chan_sccp and chan_skinny (one thing is that
I remember with chan_sccp is that the VM button didn't work and trying
to answer multiple incoming
Hi
I am only interested in using SIP, i would like to setup simple call forwarding either by thephone or Asterisk, it is a commonlegacy PBX feature and i am sure it is available from Asterisk.
What would be the best way to do it?
Is it hard coded into Asterisk?
Why do the SIP Cisco 7940's call
Calling bv from pstn phones my log shows correct caller id number and name
Calling from voipjet with cid set to 10 digit number(207826) my logs
show bv adding a leading + and setting the name to egypt as a result
Everything else I call from voipjet gets caller id correct
Could be they
Is it possible to define an extension that is not allowed to make or
receive calls, unless an agent logs in? Obviously it would require
that the extension be able to dial the # for the agent to log in.
Thanks,
Waldo
___
--Bandwidth and Colocation
Yeah, in your zapata.conf just give each channel a different context
setting.
It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to
use the
zapata_custom.conf file, instead. You also need to use the
extensions_custom.conf file, too, though there might be a better
Asterisk don't running, because show this message
WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls'
already registered (or something close enough)
== Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
Asterisk don't running, because show this message
WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls'
already registered (or something close enough)
== Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:
I ran a trace on your TG. I see
USB phone and NAT - What has USB Phpne got to do with NAT?
USB Phone is just a hardware piece that pipes the audio output from your
softphone.
Your softphone has to take care of that.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Callum
Have you tried placing a Wait(1) before Answer() in your dialplan?
On Friday 16 September 2005 11:23, J Thomas wrote:
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id.
I was wondering if anyone knows why when I register a user
agent like XLite with Asterisk I am noticing that the
port assignment on the sip show peers command shows the port to
be different than any of the other user agents. The other user agents are
logging in from different networks from
You could name your peers by the full e.164 number associated with them
then parse the area code from that to append to the number they dialed
as a 7 digit number possibly.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Schulte
Sent:
John covici wrote:
OK, I wonder if I have something wrong -- I have the *1 in my
features.conf for the one touch record -- now I called a number, and
when the call was answered flashed the hook and pressed *1 and went
back tothe call, but nothing happpened. I am using CVS from 8/26 --
is this
I'm fighting with this right now and I'm hitting a serious frustration
point - right now all incoming calls are getting handled by the
from_pstn context which is how it honestly should be according to the
current conf files. However when I change the context from from_pstn
to aa_1 and aa_2
I had sone trouble with this also.
But I got it.
I used the latest SpanDSP
0.0.9pre1
I used the applications app)_txfax and app_rxfax from the previous release
0.0.2pre8
I have included my makefile and the two app fiels that complied on the Latest
CVS (last night) on FC3.
Apps/MakeFile
#
#
I am using alsa with asterisk.
The asound.conf is below.
When I start asterisk with /etc/asound.conf present I get errors on
the console that:
chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument
If I remove the asound.conf asterisk starts up and works. However
I NEED the
Hello,
is there a best practice to upload queue_log file into MySQL? or - better
- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
Thanks
l.
--
Assum est, versa et manduca.
___
--Bandwidth and Colocation sponsored by
Asterisk don't start, because show this message:
Sep 16 17:04:59 WARNING[1075709024]: cli.c:702 ast_cli_register: Command
'showparkedcalls' already registered (or something close enough)
== Parsing '/etc/asterisk/features.conf': Found
-- Added extension '700' priority 1 to parkedcalls
Hi, im working about sip call pick and *8
works very fine but I pickup ringing phone on the same group. What happen if I have
more than one ringing call?
I tryied *8#exten, *8eten# but it doesnt wotk.
Is it correct? How it does work ?
Thanks
Giordano
As was already suggested, Wait() is your friend.
We had the same problem when our PRI was installed. It was supposed to
include Caller ID Name delivery, but it seemed to be hit-or-miss as to
if it would work.
This is what I found: When people call our auto-attendant and dial an
extension,
On CVS head there is
app_directed_pickup
It will let you pickup a ringing extension directly
without having to worry about pickup groups etc.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano
GrandisSent: Friday, September 16, 2005 11:19 AMTo:
This is because Caller ID name is being send in the FACILITY messages
instead of in the SETUP.
Wait(1) is the solution, There is nothing wrong with your install or
asterisk.
Asterisk picks up on SETUP as it should. When it recieves the Name via
FACILITY it propigates it in the channel,
Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1
distributions (I'm running FC3 linux on an Opteron 2 processor
system)? Are there any patches out there to make it work?
gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX -
DREAD_MMAP -DOSS -DTERM_CONTROL
We just uploaded the latest and greatest version of the idefisk iax2
softphone, version 1.24
Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php
Changes since the last release include:
- history panel is working
- receiving messages and urls (sendtext command in
On Fri, 16 Sep 2005, J Thomas wrote:
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
I cannot use CVS, is
there anoyher way to use direct pickup ?
Thanks again
Giordano
Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Alexander Lopez
Inviato: venerdì 16 settembre 2005
17.53
A: Asterisk Users Mailing List -
Non-Commercial Discussion
Oggetto:
see if it compiles into ur install
Iqbal
Giordano Grandis wrote:
I cannot use CVS, is there anoyher way to use direct pickup ?
Thanks again
**Giordano**
*Da:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *Per conto
Thanks -- it works very nicely -- I will have to try that filename,
but otherwise it seems to be just fine.
on Friday 09/16/2005 Kevin Bockman([EMAIL PROTECTED]) wrote
John covici wrote:
OK, I wonder if I have something wrong -- I have the *1 in my
features.conf for the one touch record
Hi,
I have 2 asterisk boxes as Gateway, in this arrangement.
(PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE)
everything works great, in both directions (receiving and making calls),
but when i get a call on the (ANALOGPHONE), I haven't been able to
transfer it to
On Fri, 2005-09-16 at 09:36 -0400, David Sampson wrote:
If I understand correctly you are supposed to patch the Makefile in the apps
directory and then run the main Makefile. I've tried both ways - the patch
failed on the main Makefile. Should I try to make that work?
There's next to
Best to log directly to MySQL. Add in ODBC code.
You are not the first to ask for it.
-bill
On 16-Sep-05, at 11:06 AM, lenz wrote:
Hello,
is there a best practice to upload queue_log file into MySQL? or -
better
- to have Asterisk log the queue_log straight to MySQL?
Is it worth doing?
PA-122TI from www.broad-tel.com supports T.38 and Fax pass-thru.
On 9/15/05, Rosario Pingaro [EMAIL PROTECTED] wrote:
about spa-2100, the t38 stream is on UDPTL and so asterisk passthroughdoesn't work.- Original Message -
From: Nenad Radosavljevic [EMAIL PROTECTED]To:
Hi,
Has someone figured out how to change the main
autoattendant message easily?
Right now, you call *77 and record the message.
Then you have to get into the Unix/Linux command
line to get that message over to where it will
be used. Is there a simpler way?
Thanks for your help.
Maybe I should ask this question that I know has been
discussed to death.
"stable" = 1.0 release
"CVS HEAD' = 1.1 release
Is this a correct statment
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
SampsonSent: Thursday, September 15, 2005 12:17 PMTo:
What you were trying to do, *8#exten, is almost right I think. Look at
it like this instead, though. The # is a pickup group number:
*8x
where x is the pickup group you want to pick up a call from. I could be
wrong but that's how I understood it.
Mojo
Giordano Grandis wrote:
I cannot
When I have voip conversations over asterisk through my computer the voice
quality is nice and loud and quite clear. When I go through my Sipura 2K
then the conversations are typically very muted and my responses sound somewhat
delayed. I've tried fiddling with settings under asterisk and the
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to
start:
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 16 20:36:51 ERROR[6750]:
I'm unable to connect to voip-info.org... Anybody else have the same
issues, ro is it just me?
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the
This is an excerpt from the log file,
My problem is that randomly, 1out of 3 or 1 out of 2, some calls are
not going out and this is the message in the log file,
The device that should provide the frame is a Sipura 3000 which has its
FXO providing outside connectivity,
24185 Sep 16 10:35:40
I got right in just fine...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters
Sent: Friday, September 16, 2005 10:26 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] wiki down?
I'm unable to connect to
Raise both gains from -3 to 5 that solves volume problem,
log in, click admin, advanced, I guess is on the sip tab,
Matthew Harrell wrote:
When I have voip conversations over asterisk through my computer the voice
quality is nice and loud and quite clear. When I go through my Sipura 2K
then
Noah Miller wrote:
Hi Andres -
The two that we have are just used as lobby phones. They're good
little phones, but if you have the money, I'd definitely recommend
the IP501 instead. The screen is MUCH better, and having full
speakerphone is great! Plus the 500/501 just feels a little
Dear all,
Anyone get the lastest spandsp with udptl.c and tpkt.c
compile in Fedora 3?
tpkt.c: In function `accept_thread':
tpkt.c:140: error: `TCP_NODELAY' undeclared (first use
in this function)
tpkt.c:140: error: (Each undeclared identifier is
reported only once
tpkt.c:140:
On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote:
Are there drivers available for Xten Softphones?
Do they work with X-lite/Eyebeam Softphones?
And work on Linux (kernel 2.6)?
On 9/16/05, Bill McCready (PCPhoneline.com)
[EMAIL PROTECTED] wrote:
Hi all,
Thanks, is there a standard schema for queue_log or can I define it myself?
Thanks
l.
In data Fri, 16 Sep 2005 18:48:13 +0200, William Lloyd [EMAIL PROTECTED]
ha scritto:
Best to log directly to MySQL. Add in ODBC code.
You are not the first to ask for it.
-bill
On 16-Sep-05, at 11:06
Hi Andres -
I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP
over asterisk, and yes, sip are 100% tweakable,
how do you configure your system, all by hand?
Yeah, by hand. When I first started doing this there was no such
thing as AMP. Plus, I've got some
Hello Everyone,
For regular call pickup you can't really specify a pickup group number...
that's why it's set in the configuration.
For directed call pickup you need to have the latest CVS head as it uses an
API call that Kevin put in espically for me to use lastnight.
Joshua Colp
Hi,
I noticed this weird behavior - in my office I use mixed phone technology. I
use Sip and Zap phones, analog and ISDN. I also defined a pickup feature and
everything works prima to the time when I want to pickup call with ISDN phone.
The console says (when I press my pickup extension *6)
I am trying to get asterisk to MIX micely with alsa or oss.
Anybody doing that? if so can you share with me how you did it.
When I set modules.conf to load alsa and noload oss asterisk
starts and binds the sound port. nothing else will play.
If I stop asterisk other things play.
If I put
I am going to be traveling and I wanted to be able to get on the
internet and call thru * to make calls. The problem is I do not have a
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA
Bob
___
--Bandwidth and Colocation
dyndns.org
From: Robert Rawlinson [EMAIL PROTECTED]
Date: 2005/09/16 Fri PM 03:51:56 EDT
To: Asterisk asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to access * thru router when ip address is not
known
I am going to be traveling and I wanted to be able to get on the
anyone with some info on this?
thanks again.
On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote:
Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider, all in g729 will require to
purchase codecs from Digium?
also, in this scenario the
On Fri, September 16, 2005 19:53, Wiley Siler said:
I got right in just fine...
W
Me too now. :-/
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to
Hello Erik!
check out this website:
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
i have both codecs running on gentoo without problems(working with cisco 7960
and snom 190). regarding your hardware question i don't have enough experience
yet, sorry.
regards
christian
On
voip-info is down from time to time, guess more spending for their server
hardware is needed.
On Fri, 16 Sep 2005 19:26:18 +0200 (CEST)
Francesco Peeters [EMAIL PROTECTED] wrote:
I'm unable to connect to voip-info.org... Anybody else have the same
issues, ro is it just me?
--
Francesco
Assuming your Asterisk mail client is set up correctly, cron this twice a
day on your Asterisk box:
ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address
[EMAIL PROTECTED]
Check your mail before you call and verify that the IP address has not
changed, if it has, modify your client
There is a standalone linux version of xlite available on their
homepage.. http://www.xten.com
On Fri, 2005-09-16 at 11:38, Guillermo Salas M wrote:
On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote:
Are there drivers available for Xten Softphones?
Do they work with X-lite/Eyebeam
whoops, should have prefaced that with the Asterisk box has to be forward of
any firewall, otherwise it's going to return a 10.X.X.X or 192.168.X.X IP
address.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, September 16, 2005 2:28 PM
To: 'Asterisk Users
Hello,
Is anyone successfully working with iPlan Networks in Argentina for
telephony service?
I'm interested in hearing about people's experience with their service
and support.
Regards,
Ilan
___
--Bandwidth and Colocation sponsored by Easynews.com --
Hi, your project is indeed interesting, however for learning purposes
i do need to know the answer of at least:
1- Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider(g729), all in g729 will require to
purchase codecs from Digium?
2- also, in
On Fri, 16 Sep 2005 16:09:37 -0500
Erick Perez [EMAIL PROTECTED] wrote:
Hi, your project is indeed interesting, however for learning purposes
i do need to know the answer of at least:
it is not my project.
1- Using sipura sip/g729 to connect to an asterisk server that will
server as a
Colin Anderson Wrote:
Assuming your Asterisk mail client is set up correctly, cron this twice
a day on your Asterisk box:
ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address
[EMAIL PROTECTED]
Check your mail before you call and verify that the IP address has not
changed, if it has,
Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to
work with the Cisco 79* series phones?
I'm not sure if the are the statndard POE or not
-Darren
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Is there a linux sip or iax phone that will autoanswer
and connect to the
console or soundcard?
I found linphonec but it does not autoanswer from what I can tell.
Jerry
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Why do you need to compile it?
Isn't it available as an rpm package?
--
#Joseph
On Fri, 2005-09-16 at 08:56 -0700, [EMAIL PROTECTED] wrote:
Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1
distributions (I'm running FC3 linux on an Opteron 2 processor
system)? Are there
Hi Hatton,
Could you provide some examples of the config files for this. Im trying to
do the same. Im confused with some of the other posts (its not hard to
confuse me!) Some say its just the zapata and some say theres way more to
it.
I have 4 FXO ports, 2 on one number and 2 on another and want
Another method comes to mind similar to my last posted method, but for
*behind* a firewall if you have a static hostname but dynamic ip (lots of
ISP's do this):
traceroute -m1 my.statichostname.net | mail -s My Asterisk IP address
[EMAIL PROTECTED]
Yet another method is, a lot of Linksys / SMC
I came up with a solution a few years ago that only required a web
browser to get the current IP address.
It requires an account with ssh access allowed on a web host.
1) Install your public ssh key on the account where the web pages live.
2) If you are concerned about others getting your IP
Joseph wrote:
Why do you need to compile it?
Isn't it available as an rpm package?
I will assume he knows why he needs to compile it.
See if the source for the rpm, deb, or whatever from the distro you are
running will build for you. That will often get your system to the point
where
I've tried to google this issue with no resolution.
I'm having the same issue as this person:
http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html
Basically, anytime I try to dial out on my TDM400P w/ FXO, I get we're
sorry, but your call cannot be completed as dialed.
When
The way I accomplished this is to leave my default context empty, then
define the distinctive ring in Asterisk, but send it to the empty
default context.
Asterisk will generate a warning that it doesn't know what to do, but it
will also do nothing with the call.You can just do this with
Where do I find or what is the default password for a GrandStream BT 101
for the web interface
___
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joshua
Abbott
Sent: zaterdag 17 september 2005 1:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Grandstream
Where do I find or what is the default password for
- Original Message -
From: Christian B [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, September 17, 2005 5:44 AM
Subject: Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
On Fri, 16 Sep 2005 16:09:37 -0500
Erick Perez [EMAIL PROTECTED] wrote:
Hi,
It seems like my ATA is making a ringing noise... (as it used to), but
now (After the upgrade from 1.0.7 to 1.2) asterisk also is either
making the ringing, or passing the PRI ringing from the telco on to
me. Any suggestions on how to fix this?
___
I have 4 FXO ports, 2 on one number and 2 on another and want to have
different incoming rules\IVR depending upon channel called.
Is it as simple as changing the contexts in the zapata.conf or is there more
to it.
Here is what my experience was. Understand when reading it that I am
running
It almost seems like I'm getting an inbound ring from my PRI/IAX
terminator and asterisk is also generating a ring. If I put an 'r'
in my dial statement I get only one ring. But is there any issue to
be taken with putting an r in? And apparently I have to put a
'timeout' value in?
On
Hello all,
I have set up a free IAX calling platform similar to
FreeWorldDialup/IAXtel. You can access it at http://www.goiax.com/
The website is still very beta but it will allow you to sign up for a
virtual phone number, and you can make outgoing calls to US toll-free
numbers. There is
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