Hi,
This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thingbut don't know how to accomplish this. If you've or anyone here figured out, please let me know.
Thank you very much,
Ron
[EMAIL PROTECTED] wrote:
I am trying implement two-stage
WipeOut wrote:
Doesn't NuFone use SER in front of Asterisk? so using asterisk purely
as the PSTN gateway..
Later
Nufone offers IAX termination, SER is SIP - or am I missing something here?
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Adam Hart wrote:
WipeOut wrote:
Doesn't NuFone use SER in front of Asterisk? so using asterisk purely
as the PSTN gateway..
Later
Nufone offers IAX termination, SER is SIP - or am I missing something
here?
Sorry, I was not thinking, you are correct..
Just most termination providers (SIP
I DON'T KNOW
From: Girish Gopinath [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] H323 - SIP Interoperability
Date: Thu, 01 Apr 2004 22:46:10 +0530
Hello,
From: pesb [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 - SIP Interoperability
Date:
Hi,
I would love to see your code.
We have the same problems with GS phones.
Bart
- Original Message -
From: Brancaleoni Matteo [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, April 03, 2004 4:35 AM
Subject: Re: [Asterisk-Users] cron job to reboot GS101
sure, we do that with a
I am interested to learn if I need to have ztdummy installed if I do not
have any zaptel hardware in my machine?
I have found a lot of references with RTP problems which were related to
RTP timing (or lack of it).
My problem is that voice coming from SIP hardware is OK, but voice going
from
Bartosz Jozwiak wrote:
Hi,
I would love to see your code.
We have the same problems with GS phones.
I don't have a GS phone yet so I don't know exactly, however I assume
you could do it with a simple call using lynx from cron...
--
Best regards,
Duane
http://www.cacert.org - Free Security
On Fri, 2 Apr 2004, Steven Sokol wrote:
I don't believe it is making it to Asterisk. I have tried changing the SIP
Proxy IP field to both the IP address and the host name of my Asterisk box
to no avail. I never see so much as an attempt to register. Weird, huh?
Any further thoughts would
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(w) 256.859.4508
(c)256.655.0321
(iax) 700.859.4508
Ask me about Asterisk
- Original Message -
From: Steven Sokol [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 9:50 PM
Subject: RE: [Asterisk-Users]
- Original Message -
From: Steven Sokol [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 9:50 PM
Subject: RE: [Asterisk-Users] WiSIP Firmware Version F?
I don't believe it is making it to Asterisk. I have tried changing the
SIP
Proxy IP field to both the IP
G'Day,
I have a bit of FireFly problem that hopefully someone has seen before.
What happens is if I make to or receive a call from the FireFly network
the call will connect successfully. However, around 10 seconds after I
answer the call I am disconnected. The weird thing is same thing happens
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But,
you have to log in to the web interface before doing this.
If I were motivated to develop this for a cron job I would look at using the `wget`
program and the --spider option.
Walker
--
Walker Haddock wrote:
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone. But, you have to log in to the web interface before doing this.
lynx has username/password options, well unless they used sessions...
--
Best regards,
Duane
http://www.cacert.org - Free
on your asterisk box:
tcpdump port 5060
Ok. Got it. Pulver had the outbound sip proxy port set to something other
than the standard SIP port (5060). I changed it registered just fine.
One other question. Do you get caller id with incoming calls to your WiSIP?
I don't seem to get
WipeOut wrote:
Doesn't NuFone use SER in front of Asterisk? so using asterisk purely
as the PSTN gateway..
Absolutely not. We run nothing but Asterisk.
Jeremy McNamara
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[EMAIL PROTECTED]
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/03/04-10:19:04\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
See :- http://lists.digium.com/pipermail/asterisk-users/
2016-May:[ Thread ] [ Subject ] [ Author ] [ Date ]
[ Gzip'd Text 745 bytes ]
2007-November:[ Thread ] [ Subject ] [ Author ] [ Date ]
[ Gzip'd Text 2 KB ]
Regards...Martin
--
Please avoid sending me Word or PowerPoint attachments.
On Sat, 3 Apr 2004, Steven Sokol wrote:
One other question. Do you get caller id with incoming calls to your WiSIP?
I don't seem to get anything.
It seems to work here. I get asterisk for many calls but I am sure that
is because I some fine tuning to do.
--
Andrew McRory - President/CTO
On Apr 2, 2004, at 7:14 PM, Hall, Eric M. wrote:
I'm starting to get this to work! Well I got Voice Mail to work!
All calls goes to voice mail without ringing the users phone (iaxComm).
Here is my iax.conf and my extensions.conf
Any help would be great!!
I don't see anything really obviously
Now its not even going to voice mail.. Here is the output from the debug
[EMAIL PROTECTED] asterisk]# asterisk -r
== Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk CVS-03/31/04-12:57:49, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
Martin wrote:
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/03/04-10:19:04\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
Hi!
I am interested to learn if I need to have ztdummy installed if I do not
have any zaptel hardware in my machine?
No, not necessarily. You'll only need it if you want to use MeetMe
conferencing. Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
I have found a lot of
On Saturday 03 April 2004 10:36, Martin wrote:
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/03/04-10:19:04\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\
Title: No ringback
I just configured Asterisk on a new machine, and other things seem to be working fine, I don't get any audible ringback when dialing calls from a SIP phone or a standard phone connected through a TDM400P. What am I doing wrong here?
Thanks
-brian
How'd your trip home go?
You had me worried to death the other night; in the infamous words of my
father's brother, You looked like you'd have to get better to die.
Anyways, hope the trip back was OK, and will be in touch soonly
regarding that freakin' stuck 800 number.
Thx.
B.
That works well! Thank you very much. :)
Wade J. Weppler wrote:
I was able to get the extension/channel problem fixed. In version .03,
change the following lines, starting at line 329:
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[EMAIL PROTECTED]
Title: No ringback
I had a similar problem. What I did what
checked out the version before 03-02-2004. Some change after that date is
causing the problem.
Gene
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie
Sent: Saturday, April 03, 2004
4:32 PM
Hi all;
I think I have the capacity issues figured out. My next question is
whether I can use asterisk for a redundant solution so that if any
hardware failure occurs on the phone switch, a spare PBX can route the
new calls. I have not been able to find this in the docs, and IIRC, it
is
You had me worried to death the other night; in the infamous words of my
father's brother, You looked like you'd have to get better to die.
... so your uncle?
-A.
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Title: No ringback
Thanks. Actually,I got the latest from the cvs
repository and it's fixed there, too. I suspect that it got broken at some point
briefly before someone fixed it.
-brian
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gene
KochanowskySent:
curl could also be used. Since people asked I'm going to write it up
tonight since I use a GS as well until my Cisco shows up.
On Sat, 2004-04-03 at 09:52, Duane wrote:
Walker Haddock wrote:
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone.
But, you have to
Dunno why your phone isn't allowing you do negotiate
g711u but I can tell you how to upgrade the firmware. I
called them on Thursday for myself and they gave me the
following tftp server address for which to program my
phone.
4.3.153.50
Load this into your phone's tftp area and reboot it.
It'll
Hello-
I am in the process of adding a new provider to my asterisk box (both
for outbound termination as well as inbound DID). They are going to be
delivering and receiving traffic via SIP only.
Now, in IAX via Voicepulse or others I know that I can simply have one
registration statement
No matter what I try, Asterisk won't let me out of the console. If I
CTRL+C, of course, the process will terminate.
I started asterisk like so:
/usr/sbin/asterisk -gc
and here's what I get when I 'exit':
*CLI exit
The QUIT and EXIT commands may no longer be used to shutdown the PBX.
What happens when you do stop now like the error states?
Sean
-Original Message-
From: Ryan Parlee [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 03, 2004 9:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unabled to exit console
No matter what I try, Asterisk won't let me out
No matter what I try, Asterisk won't let me out of the console. If I
CTRL+C, of course, the process will terminate.
I started asterisk like so:
/usr/sbin/asterisk -gc
Looks like Asterisk wasn't started as a daemon. Just run
/usr/sbin/asterisk. Then to connect to asterisk run
I found some old messages regarding a possible pkt8 DTA bypass. Anyone
is using Packet8 with Asterisk?
==
http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat
Got Softphone Working with Packet8
Friendly name: {Anything you'd like.}
SIP domain: packet8.net
SIP proxy:
Okay, but if I do
/usr/sbin/asterisk
Then when I connect, using -r I don't get debugging information. Isn't
there a way to make asterisk deamonized and still get the - stuff when
you reconnect?
Thanks, Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Then when I connect, using -r I don't get debugging information. Isn't
there a way to make asterisk deamonized and still get the - stuff
when
you reconnect?
Type: set verbose 3
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
Hi,
Where could one find this (op_server.pl)? I searched the wiki and google...
But did not find it.
-G
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino
Sent: Friday, April 02, 2004 12:37 PM
To: [EMAIL PROTECTED]
Subject: Re:
to start daemon type:
safe_asterisk
to start console type:
asterisk -vvvr
(notice the r at the end)
Note you don't really need that many v's just more for more verbosity
Matt Riddell
Then when I connect, using -r I don't get debugging information. Isn't
there a way to
just do -vvvr
-Original Message-
From: Ryan Parlee [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 03, 2004 11:39 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Unabled to exit console
Okay, but if I do
/usr/sbin/asterisk
Then when I connect, using -r I don't get
I'm using packet 8 with a standalone handset here in Australia, haven't
tried to run it via my asterisk server yet.
I have to say I'm pretty impressed with the flat rate service, the
quality could be a bit better but for a flat rate who cares.
Cheers,
Dean
-Original Message-
From:
We have 10 Cisco 7960 phones at our office and a single static IP. Our
asterisk server sits in the colo facility at our ISP. All phones are setup
with a unique voip_control_port and they are all able to dial out. However,
my phone is the only one that can receive a call.
Every phone in the
If I do this:
1) /usr/sbin/asterisk[or]/usr/sbin/safe_asterisk
2) /usr/sbin/asterisk -r
It works, kind of, but it doesn't have near the detail as when I do:
3) /usr/sbin/asterisk -gc
and it's not in pretty colors like it is in (3).
Oh well, I can always stop and restart when
Can you post some of your sip configs and your extension configs.
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Parlee
Posted At: Sunday, April 04, 2004 12:10 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users]
Sure.
; sip.conf ;;;
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[gateway]
type=friend
host=111.111.111.111
; [EMAIL
Okay, I finally figured this out. For some reason these phones are smart
enough to maintain their own hole in the firewall. When I added all the PAT
entries into the firewall everything works without a hitch. Now I can
concentrate on added more features to my dial plan.
Thanks,
Ryan
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