Hello *'s,
i am using Realtime Sip drivers but its not working here is my configs:
extconfig.conf
[settings]
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is
Hi,
I have had problems with receiving faxes with rxfax.
Using bristuff with four hfc cards and spandsp 0.2.0 pre6 I got the
symptoms of frame slips.
I've tried all the debugging tips in this thread.
I've tried moving from kernel 2.4 to 2.6.7.
The only symptom of something wrong has been the
Hey,
I've just installed hardened gentoo with selinux and emerged the selinux policy's for asterisk and emerged asterisk after it, now whenever i want to run asterisk i get:
Dec 31 11:56:46 WARNING[4248]: manager.c:1474 init_manager: Unable to bind socket: Cannot assign requested address
and
On Fri, 2004-12-31 at 11:59 +0100, christophe de coninck wrote:
Hey,
I've just installed hardened gentoo with selinux and emerged the
selinux policy's for asterisk and emerged asterisk after it, now
whenever i want to run asterisk i get:
Dec 31 11:56:46 WARNING[4248]: manager.c:1474
Thanks for noting me, it was late yesterday and early this morning ;)
just checked out manager.conf and there was the fault, thnx
On Fri, 2004-12-31 at 12:32, Steven Critchfield wrote:
On Fri, 2004-12-31 at 11:59 +0100, christophe de coninck wrote:
Hey,
I've just installed hardened gentoo
Sirs,
According to RFC 2705 (MGCP), these are the parameters that are used in the
transactions:
ReturnCode,
Connection-parameters
-- DeleteConnection(CallId,
EndpointId,
ConnectionId,
Hi,
I do not understand the difference between SIP and IAX, is it only two
different protocols or something more special.
The problem I have is that I've created two users
Aix.conf
register = users1:passwd1
register = user2:passwd2
[user1]
type=user
context=default
secret=passwd1
I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1
(see config below) and with a bit of
messing about using sample config, have been able to make the test call to
device 1000, and also through to the IAX
test number at Digium. However, operation is extremely flaky - I
Might be related to the musiconhold files using different encoding rates ?
Just an idea, also a newbie :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paid Up
Sent: vendredi 31 décembre 2004 14:01
To: asterisk-users@lists.digium.com
Subject:
Hi,
1) 0.0.0.0 just means listning on all interfaces and their ip adresses, not
a problem.
2) Do a set verbose 100 to see if you have any communication with the sip
phones or startup asterisk with asterisk -vvvggg
3) This is because a MPG3 file used for music on hold isn't support or that
On Tuesday 28 December 2004 04:32, C F wrote:
Just a note on this. I tried using an external device with the TDM400
configured as 4 FXO to ring even with asterisk. But no matter how I
configured it, asterisk always picked up. and the external device
didn't ring (just the first ring for
Now I have searched around and not seen anything to do this.
I want to in remote locations were we need to have single or 2 PSTN
lines for in dial as little hardware as possible and as stable as
possible so that they will operate without user intervention.
What I want to do is be able
Hi,
What do you think that the problem might be if a program has a segmentation
fault at the same library call? The library call is from libpthread.so.0
and the call itself is pthread_mutex_locl ( ). I have enclosed the core
dump information below. The program comes up and then does the
Checkout http://www.mediatrix.com (FXO device 1204) or
http://www.multitech.com.
I have been looking into this myself. It appears that Nortel has an
arrangement with Mediatrix and uses these devices where a remote FXO is
needed that would be cost prohibitive to put in a full chassis. Avaya
Hi,
This is a thank you message for all that helped me
including Max from www.asterisk-support.ru with whishes
of a Happy New Year.
Althought I still have a problem I'm happier I've
50% of my task complete. I'm using two TA from Draytek (router 2600V
/router 2500V) 3 ADSL lines (2 for TA
IAX2
- Original Message -
From: Serge Schumacher [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 31, 2004 8:00 AM
Subject: [Asterisk-Users] IAX users
Hi,
I do not understand the difference
Sorry ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: vendredi 31 décembre 2004 16:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX users
IAX2
- Original Message -
From:
SIP is a XML-like control channel and is used to negotiate a separate RTP
channel which carries the audio. It is complicated to set-up in cases of
firewalls and NAT, but is an open standard.
IAX2 is a candidate open standard and merges all traffic onto a single UDP
stream - control and audio
Checkout http://www.mediatrix.com (FXO device 1204) or
http://www.multitech.com.
I have been looking into this myself. It appears that Nortel has an
arrangement with Mediatrix and uses these devices where a remote FXO is
needed that would be cost prohibitive to put in a full chassis. Avaya
Use res_config_odbc ... Lots of people have problems with the mysql one. I
have never once had a problem with the odbc one. The wiki even has a small
how to on setting it up.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of adnan
There are two different approaches available:
1. Hardware
What you want is a remote SIP gateway. These are boxes which have
FXS/FXO/EM ports in some combination on one side and and an ethernet port
on the other. Most of these boxes were originally designed to run H.323 and
had SIP firmware
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold
by BroadVoice work with Asterisk or is it a locked down device like the
Vonages ATA186?
Adi
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On Fri, 31 Dec 2004, Serge Schumacher wrote:
extensions.conf
exten = 550,1(Dial,IAX/user1);
exten = 551,1(Dial,IAX/user2);
and the error I get :
Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No
application 'IAX/user1)' for extension (default, 550, 1)
== Spawn
Hello,
I've done a very straightforward install of Asterisk, and can't seem to
get it started.
This is a proof-of-concept installation, and currently does not have
any T1/E1 or FXO/FXS cards in it. I just want to use it as an internal
SIP server for now.
However, when I try to start
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe
These are messages from mpg123 NOT Asterisk.
bkw
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The RFC specification alone is not sufficient, there are many signaling
packages that are defined elsewhere.
Also, RFC 2705 is out of date, see RFC 3435
Leonardo J. Tramontina wrote:
Sirs,
According to RFC 2705 (MGCP), these are the parameters that are used
in the transactions:
On Fri, 31 Dec 2004, Adi Linden wrote:
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold
by BroadVoice work with Asterisk or is it a locked down device like the
Vonages ATA186?
You'd probably have to ask them that. Just so you know, you can buy that
phone elsewhere. It
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold
by BroadVoice work with Asterisk or is it a locked down device like the
Vonages ATA186?
You'd probably have to ask them that. Just so you know, you can buy that
phone elsewhere. It is made by Pulver Innovations
I didn't look into any disconnect fees yet. That's a good one to be aware
of since the phone appears to be available in a bundle with their service.
I had a look at the Pulver cordless. How does it (at $199) compare to the
Zyxel 2000W (~$250 from voipsupply.com)?
Adi
They are the same phone.
Thanks to your generous donations, people all over will continue to
hear such great programming as Fresh Air, All Things Considered, Morning
Edition and the BBC World News...
Sorry, I meant to say - such great programming as 'Barn', 'Food
Services', 'Security' and 'I.T. Services'
Thats
you have two 'friend' entries in your sip.conf. it uses the second,
which is not what you want. one should be peer and the other user.
though a number of versions of asterisk don't actually work with
peer/user, a major pita.
so try reversing the order of the two entries if you have problems
A friend of mine urgently needs a skilled asterisk, linux, Mysql type person
for a one off job. Preferrably London based.
Email me with your details if your interested.
Charles
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Hi ALL;
In IAX protocol, both rtp and signaling are handled
on the same port, so the Asterisk is always in the path of rtp
traffic.
Am I right?
If yes, is there anyway to set Asterisk just as
signal proxy ?
Regards
Mohammad
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Anyone have a example of how to setup RDNIS in *?
To date we have been giving each voicemail user a individual DNIS but would like
to consolidate all the numbers into one and just use RDNIS to route the
call.
Thanks,
Gary
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Asterisk-Users
Hi,
I'm playing with the agent/queue system. Everything work well with v1.0.3.
but I want the 'Action: Agents' in the manager API that is only on the CVS
version. So i switched to, but now the Queue/Agent system barely work. (my
agent don't get the call)
Where I can get a 'stable' CVS version?
Hi ALL;
Hi Matthew;
Can we integerate ASTCC Sipfriend or Iaxfrien
tables with Mysql-Realtime driver?
Regards
Mohammad
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To
Whisker, Peter [EMAIL PROTECTED] writes:
SIP is a XML-like control channel [...]
I think you meant HTTP-like. :-)
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
If you look through the astcc-admin source a little, you should find it
quite easy to modify astcc to use the realtime drivers. If you get it
changed, please submit it as a bug report. That is something that needs
to get changed.
Darren Wiebe
[EMAIL PROTECTED]
mohammad wrote:
Hi ALL;
Hi
double post
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How about Technical Services?
On Fri, 2004-12-31 at 13:32, Alspach Family wrote:
Thanks to your generous donations, people all over will continue to
hear such great programming as Fresh Air, All Things Considered, Morning
Edition and the BBC World News...
Sorry, I meant to say - such
I am looking for a German language softphone. Is there such a thing?
Adi
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I tried Firefly in WinXP and it works fine. I tried it on Win98, it looks
it up. Anyone experience this? Any ideas?
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 12/30/2004
All,
I have FC3 fedora core 3 and just installed and compiled 2.6.10.
after rebooting I attempted to recompile zaptel-1.0.3. I did a make clean
then make. I got the following errors.
Any suggestions?
---
/usr/src/digium/zaptel-1.0.3/torisa.c:1139: warning:
Hi. I can't figure this one out. Hope someone can help me.
[EMAIL PROTECTED]:# cat /etc/odbc.ini
[Asterisk]
Description=PostgreSQL asterisk
Driver=PostgreSQL
Trace=No
TraceFile=/tmp/odbc.log
Database=asterisk
ServerName=localhost
UserName=
Password=
Port=5432
Protocol=7.4
Sorry about this. Just figured it out.
In res_odbc.conf its supposed to be pre-connect and not preconnect.
On Saturday 01 January 2005 02:30, Arthur B Olsen wrote:
Hi. I can't figure this one out. Hope someone can help me.
[EMAIL PROTECTED]:# cat /etc/odbc.ini
[Asterisk]
I have a stable server and want to upgrade. How do I upgrade to the
latest version of * ?
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When I try to start up zaptel, whilst running ztcfg, I get the following
error:
Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or
address (6)
My /etc/zaptel.conf is:
fxsks=1
fxoks=2
loadzone = au
defaultzone=au
Channel 1 is a X101P card connected to the PSTN and
We try a X-Lite client from remote to connect to my *
I can call X-Lite and X-Lite can call me. However, X-Lite can hear my
voice, while I cannot hear him.
*CLI shows
*CLI (date) NOTICE[4637] rtp.c:317 process_rfc3389: RFC3389 support
incomplete. Turn off on client if possible
RFC3389: 5 bytes,
i am facing unusual and wiered error in asterisk using Realtime MYSQL
driver . Asterisk runs well and smoothly with absoulutely no error or
warning but everytime i power-on my sip-phone ,booting, initializes
and then asterisk suddenly quit with the error.
_*Segmentation Fault (core
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