My asterisk server(s) are behind NAT, and I am a customer of Vonage
(thrice-over), iconnecthere, and Net2Phone.
There are still some rough edges (especially with iconnecthere) but
overall it is not correct to say that they won't work.
B.
Thats great to hear. Can you please share your
rnc Info Lists wrote:
Thats great to hear. Can you please share your config files that connect
iconnecthere and net2phone via SIP? I think there are a number of people
here who have tried and not been able to get it to work.
Here's what I'm using for iconnecthere. They provide me with both
costas wrote:
I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal.
1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event
I have now is music on hold
I have installed ztdummy
MOH gives me this message now Read 372 bytes of audio while expecting
1600
And no sound
If I run modprobe ztdummy I get no sound
Even the welcome message etc do not work
And the error message changes to Read 372 bytes of audio while
hi,
i am trying to do autoattendant but failing. as in the
manual i inserted the background(welcome-mainmenu)
file so that after the sound the caller can dial the
extension he wants to call. i figured that the
background sound wasn't coming in the asterisk. how do
we do this without first loading
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I am using [EMAIL PROTECTED], all the SIP traffic will be sent to
iptel.org proxy and the proxy will take care of NAT traversal.
Hi,
I would love to participate in your test.
We have several * machines.
Please let me know further details.
Ta
Senad
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Can anyone give me presise instructions on how to compile cdr_mysql.so?
When I initially installed asterisk on the system, I didn't have mysql
installed. Since then I have installed mysql, created the database and
table structure for cdr_mysql and placed the appropriate settings in the
[EMAIL PROTECTED] wrote:
Can anyone give me presise instructions on how to compile cdr_mysql.so?
When I initially installed asterisk on the system, I didn't have mysql
installed. Since then I have installed mysql, created the database and
table structure for cdr_mysql and placed the
On Sat, 25 Oct 2003 [EMAIL PROTECTED] wrote:
Can anyone give me presise instructions on how to compile cdr_mysql.so?
did you get asterisk-addons ? cdr_mysql been moved there
When I initially installed asterisk on the system, I didn't have mysql
installed. Since then I have installed
I just went over and grabed the asterisk-addons directory from the CVS,
changed into the directory and executed a make install and got the
following error:
make: ***[cdr_addon_mysql.o] Error 1
You have any suggestions about this?
AJ
___
[EMAIL PROTECTED] wrote:
I just went over and grabed the asterisk-addons directory from the CVS,
changed into the directory and executed a make install and got the
following error:
make: ***[cdr_addon_mysql.o] Error 1
You have any suggestions about this?
AJ
I just did a cvs update and
Juan:
I think that we must continue with the discussion out of this list.
Te contacto por fuera de la lista.
Regards,
Gus
- Original Message -
From: Juan J. Sierralta P. [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Friday, October 24, 2003 7:50 PM
Subject: Re:
Please try a modification in /etc/rc.d/init.d/asterisk:
Replace:
daemon /usr/sbin/asterisk
with
daemon screen -d -m asterisk -vvvcng
Hope this helps.
Gus
- Original Message -
From: Alejandro Ruiz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 24, 2003 6:11 PM
Subject:
[inbound-home]
exten = s,1,Dial(${PHONE3}${PHONE4},15)
Thanks Rich, this worked like a charm, I don't know why I was thinking in
reverse -- that I would have to have Asterisk answer it to pass the ringing
to the SIP phone or I would have to force a pickup with some sort of key
sequence.
Where is the current Java binary? For that matter, where is the source for
the Java version?
Cheers,
Steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Victor
Medrano
Sent: Friday, October 24, 2003 10:34 PM
To: [EMAIL PROTECTED]
Subject: RE:
On Fri, 2003-10-24 at 21:12, David J Carter wrote:
Thanks Dave,
I can now call a sipphone number from * but get no voice throughput.
I still don't see anything coming in from sipphone though.
Dave
Have a look at rtp.conf
I have 8000 - 8060 there. by default its 1,
if you have a
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the culpret line right
before the error:
/usr/bin/ld: cannot find -lz
Any suggestions here?
AJ
On Sat, 25 Oct 2003, WipeOut wrote:
[EMAIL PROTECTED] wrote:
Install the zlib and zlib devel packages
On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote:
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the culpret line right
before the error:
/usr/bin/ld: cannot
Yes I do have gzip installed on that box. Any other ideas?
On Sat, 25 Oct 2003, WipeOut wrote:
[EMAIL PROTECTED] wrote:
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the culpret line right
before the error:
Thanks a bunch
On Sat, 25 Oct 2003, Eric Wieling wrote:
Install the zlib and zlib devel packages
On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote:
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the
costas wrote:
How did you determine that they are faulty? Did Digium replace them?
Thanks
Well when I started the device worked and over time the phone started
crackling and then randomly stopped providing dial tone and then stopped
providing dial tone at all..
I am in the UK so I
gzip is not zlib. On my Mandrake 9.2 system the zlib packages are:
zlib1-1.1.4-8mdk
zlib1-devel-1.1.4-8mdk
On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote:
Yes I do have gzip installed on that box. Any other ideas?
On Sat, 25 Oct 2003, WipeOut wrote:
[EMAIL PROTECTED] wrote:
On Sat, 25 Oct 2003 [EMAIL PROTECTED] wrote:
Yes I do have gzip installed on that box. Any other ideas?
You are looking for the the libz stuff, which if you use RedHat is a part
of zlib-devel-1.1.3-25.7 (or whatever the right number is for your
distribution).
You should be able to see it in
Thanks a lot your earlier suggestion worked. The system was lacking
zlib-devel. Now where do I insert the lines for it to load the
cdr_mysql.so since I have it built? Can you give me an exact example of
what to put here?
AJ
On Sat, 25 Oct 2003, Eric Wieling wrote:
gzip is not zlib. On my
make update, not make upgrade
On Sat, 2003-10-25 at 13:33, Rich Adamson wrote:
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the
On Sat, 25 Oct 2003, Rich Adamson wrote:
make upgrade
-- it connects to CVS and shows me all the new files it downloaded
make clean ; make install
I just tried the above, and absolutely nothing was updated. I actually
checked *.c files for dates/times, and all were from previous cvs.
Modify /etc/asterisk/modules.conf
load = cdr_mysql.so
or
load = cdr_addon_mysql.so
Then, modify your cdr_mysql.conf, like this:
[global]
hostname=localhost
dbname=astcdr
password=12jaslap3
user=amadata
sock=/var/lib/mysql/mysql.sock
Hope this helps.
Gus
- Original Message -
From:
- Create a conf file called cdr_mysql.conf in /asterisk/
[global]
hostname=localhost
dbname=asterisk
password=somepass
user=someuser
- Add load = cdr_addon_mysql.so to /asterisk/modules.conf
If you are getting errors with mysql cdr while loading asterisk, check that
you can connect to MySQL
Opps, funny how the fingers do the walking following the eyes, without
questioning the print... ;)
make update, not make upgrade
On Sat, 2003-10-25 at 13:33, Rich Adamson wrote:
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my
Thanks, got it working, Thanks to everyone!
AJ
On Sat, 25 Oct 2003, CW_ASN wrote:
Modify /etc/asterisk/modules.conf
load = cdr_mysql.so
or
load = cdr_addon_mysql.so
Then, modify your cdr_mysql.conf, like this:
[global]
hostname=localhost
dbname=astcdr
password=12jaslap3
Just submitted a patch for this on asterisk-dev.
Quick fix add the following line above line 5022 in chan_sip.c
ast_setstate(c,AST_STATE_DOWN);
Just updated to current cvs a few minutes ago primarily to get the
call pickup to function properly. Using C7960's and Snom 200 on RH9.
All
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
== No one is available to
Voicemail.conf in MySQL is not functioning where I get the following error
from Asterisk messages log file:
CLI debug output is as follows:
Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0'
I recompiled Asterisk with the aggressive echo cancellation on. That's
all I changed, honest. After recompiling, it refused to run. I tried
updating the source, etc, and eventually went back to no echo
cancellation. Every time, I got this error while starting Asterisk.
Please help! I have no
I recompiled Asterisk with the aggressive echo cancellation on. That's
all I changed, honest. After recompiling, it refused to run. I tried
updating the source, etc, and eventually went back to no echo
cancellation. Every time, I got this error while starting Asterisk.
Please help! I
Did you try to use *8 only instead of *8# ?
Last time when I tried *8 picked the call with known results
but I haven't tested any patches yet.
I really hope call pickup now works.
-- Pertti
Rich Adamson wrote:
Just submitted a patch for this on asterisk-dev.
Quick fix add the following
I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=434
requesting that Digium put up a page with links with external Asterisk
related resources. If you have a web site with Asterisk related
information, patches, samples, documentation, etc, please add a bugnote
to the above URL.
David,
We has successfully tested last week interoperability between Asterisk and
our SIP softswitch.
We can definately help you with your project. Our company is New York
based, so it will be very easy to get interconnection with you. We are mostly in
wholesale business.
Also you can get
I recompiled Asterisk with the aggressive echo cancellation on. That's
all I changed, honest. After recompiling, it refused to run. I tried
updating the source, etc, and eventually went back to no echo
cancellation. Every time, I got this error while starting Asterisk.
Please help! I have no
Think I'm a little confused on registering an iax connection; could
someone enlighten me?
I guess the real question is... when two * machines are going to rely
on an iax link (each with their own dial plan), do both machines have
to register with each other (eg, both need a 'register' statement)?
I have spotted some candidates at EBay... Problem is that Pay Pal cannot
handle payments from my country, Colombia.
Paypal isn't the only way to pay -- a U.S. Money order or wire transfer
should work just fine...
Andrew
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Asterisk-Users mailing
Questions ...
OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes,
and extensions. All exciting.
Two questions:
I'm in a natted environment and need to utilize a SIP provider to make calls
in the US. Currently I have Vonage in my natted network and it works fine,
however
Interesting. Someone thinks that a strategic use for * should be off
this list. Someone thought my FAX modem for * should be off this list.
However, nobody seems to think a 1000 messages about Grandstream phones
should be off this list.
Personally I would welcome seeing more of what people are
Questions ...
OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes,
and extensions. All exciting.
Two questions:
I'm in a natted environment and need to utilize a SIP provider to make
calls
in the US. Currently I have Vonage in my natted network and it works
fine,
Steve:
Ok, if you like to hear about Cisco BTS10200 and Cisco ITP configurations,
good... I have no problems with that...
We will discuss HERE all the configurations needed to bring up a CCS7 links
in ITP, how load a SPC formats, and how can I add an TGCP route in BTS...
Sure! Why not?
Regards,
Is IAX difficult to configure? Do you have sample configs I could look at?
Is there a rec IAX provider?
Regards,
Phillip
--
Phillip C. Jackson
[EMAIL PROTECTED]
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