4 mar 2007 kl. 21.45 skrev André Santos:
Hi,
I am implementing de Real Time architecture, I would like to know
if, their is any problem in putting the section [general] of the
sip.conf file in the table of sippeers.
You can't mix the general section with sippeers for realtime.
As these problems are very time sensitive and frustrating, I suggest you
document each change you make and do them one at a time so you can
actually know what the problem was and not introduce new problems in the
process.
Find someone who is on the phone quite a bit and will give you an
Hi all,
I install the Asterisk 1.4.1 in order to use the T.38 pass-through, but
for the moment, I cannot even make call
I have this WARNING:
[Mar 7 11:32:09] WARNING[13395]: chan_sip.c:12290 handle_response:
Remote host can't match request BYE to call
'[EMAIL PROTECTED]'. Giving
Hello,
I have problem with capi, I can't install it. I have putted all info what I
did and what I get :). I think you can suggest me how to solve this
problem.and thanking you in anticipation. I have ISDN Frtiz!Card PCI
v2.1and I want to install it to my ubuntu box (kernel:
2.6.17-10-server).
Olle E Johansson wrote:
Friends in the Asterisk community,
One thing I avoided working with for a long time is the Asterisk
voicemail code. One module in
Asterisk I've constantly been naming as one of the worst parts is
voicemail. One part of
Asterisk that I've been kind of avoiding during
I think you are mixing something here. The FritzCard is not a B1, so you
don't need the b1 modules, the firmware and the /etc/capi.conf.
You can either use the FritzCard driver (binary modules from AVM), or you
use mISDN (which is also already loaded according to your lsmod).
When using mISDN,
Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless?If yes, do you
have any suggestions for what you've done to make it work? It seems that
no amount of traffic
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It often takes 3-4 tries of entering an
extension before the system gets it right.
Below is my context that the call comes into,
John Congdon wrote:
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It often takes 3-4 tries of entering an
extension before the system gets it right.
Below is my context that
On Wed, 7 Mar 2007, Matt wrote:
Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless?If yes, do you
have any suggestions for what you've done to make it work? It
Doug Lytle wrote:
John Congdon wrote:
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It often takes 3-4 tries of entering an
extension before the system gets it right.
Below
Vidura Senadeera wrote:
Hi Team,
I have integrated asterisk with Toshiba analog PBX. NOw the live setup is
going.
Now I am facing call droping problem. It's happening ample time. 10-20
calls
are droping every day.
What could be the reason. I attached latest zapata.conf file for your
Anyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, March 05, 2007 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] TC400B
Anyone tried the digium TC400B transcoding card? What are
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
John Congdon wrote:
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It often takes 3-4 tries of entering an
extension before
Is it possible to use the include command to include other contexts if
you are using realtime for extensions? I've searched voip-info and some
people were asking about it, but I didn't find a real answer anywhere.
___
--Bandwidth and Colocation
Steve Totaro wrote:
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
John Congdon wrote:
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It often takes 3-4 tries of entering an
what is the support in asterisk for ssl voip protocols?
I am looking for a solutions to grant the possibility to some users to
use an asterisk server as a proxy voice, for talking each them in a
safe and secure mode on internet.
Is it possible?
thanks
Not sure if this is what you mean But we have includes in our
sip,extensions and voicemail files.
;#include sip.inc
We keep them commented out only because they are a copy of what is
running in realtime. Every night the include files are generated and put
in /etc/asterisk. If MySQL were to
It would be cool to get one of these and see if it can be hacked and
loaded with your favorite SIP or IAX softphone. Looking at the pic, it
looks like the dongle is both a soundcard and memory stick. Heck, I
would be glad to have it if I could get the soundcard to work.
Might as well since it
Hi Nik,
Do some googling on Phil Zimmermanns Zphone.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of nik600
Sent: Wednesday, 7 March 2007 9:38 AM
To:
I have found that relaxdtmf=yes has caused more problems than it
fixes. In my experience problems with detecting DTMF on an FXO
port
can usually be fixed by playing with rxgain and txgain.
What sort of problems have you seen it cause? I guess I could see
hitting the wrong extension
7 mar 2007 kl. 15.38 skrev nik600:
what is the support in asterisk for ssl voip protocols?
I am looking for a solutions to grant the possibility to some users to
use an asterisk server as a proxy voice, for talking each them in a
safe and secure mode on internet.
Is it possible?
No.
/O
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, March 07, 2007 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
If you just want to secure between Asterisk servers and clients that you
control, OpenVPN rules in simplicity and transparency.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
On Mar 7, 2007, at 9:58 AM, Steve Totaro wrote:
Might as well since it is free after rebate.
Just as a heads up, that rebate, like most of the others for Vonage
based items, requires Vonage activation in order to actually get the
rebate.
-chris
www.mythtech.net
Eric
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Eric ManxPower Wieling wrote:
Doug Lytle wrote:
John Congdon wrote:
Hi,
I have had an issue for a long time, and really just can't solve it.
My boss and others seem to have a problem when they call into our
asterisk phone system. It
Hi,
is it possible to have the information stored in
/var/log/asterisk/queue_log
realtime in mySQL?
thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Have a question for the group
If I have an agent is on the phone outside of the queue should that
person still get queue calls ?
Doing a show agents online I see Available however show hints I see
inuse.
Any ideas
Eric Hall
Vice-president
Amaxx, Inc.
Customized IT Solutions
I am using the * auto-dial out feature but don't want to have to specify
a channel (Zap/G2/) to connect to the extension.
Current file I use:
Channel: Zap/G2/12127778866 # I have to specify a specific
channel
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your outgoing call
I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium
cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO
modules) and a TE205P dual E1/T1 card. The problem is with the TE205P
and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf
files. I go in
Thomas Winter wrote:
is it possible to have the information stored in
/var/log/asterisk/queue_log
realtime in mySQL?
No. You need to write a custom script.
See
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
William Lloyd has a Perl script for that:
U have edited the wrong files. Can u tell us here the zaptel and the zapata
files being used by
asteriskNow 1.4-beta?
I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium
cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO
modules) and a TE205P dual
Not really what I mean. I have customer contexts, say customer1 and
customer2. I also have a LD, Local and Intl context. To allow
customer1 to dial LD, I include the LD context within the customer1
context. I want to skip text files and move to realtime for extensions
and I want to know if
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Wednesday, March 07, 2007 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue information in mySQL
Thomas
Hello,
I´m working testing every feature of asterisk in a lab. Now I am very
interested in asterisk over network mobility environment. For example : when
somebody is talking with his ip-phone ) and moving around a big enterprise,
needing to change the ip-address (other AP) would it be possible
Hi
I have a analog phone connected to my Gateway Handytone and registered to
Asterisk 1.4 I have configured my HandyTone 488
(in the section FXS Port) for make and receive calls, however I can
not transfer a call when it come via PSTN. But, when a call come from via IP
I can transfer it.
Hi Alvaro,
There was a discussion about this a little while ago. Andrew Joakimsen had
some good ideas with how to allow a wifi sip phone to roam between APs
seemlessly. His post pretty much said the following:
- Set all APs to the same channel and same SSID.
- Make sure all APs are connected
Have a question for the group
If I have an agent is on the phone outside of the queue should that person
still get queue calls ?
Doing a show agents online I see Available however show hints I see inuse.
There is a ringinuse feature for SIP devices on 1.4.X which is what you are
What Alvarion stuff are you using? Alvarion is the top rated wireless
manufactor when it comes to VOIP on there new stuff (VL 4.0 series and
B100 backhaul series). Some of there older stuff like the BrezeeAccess
FHSS will not work well because the QoS that the radio provides is
spotty at best.
Take a look at Alberto Pastor's mail on this list at 2007/02/15: moving
WiFi phone.
+jm
Alvaro Pacho wrote:
Hello,
I´m working testing every feature of asterisk in a lab. Now I am very
interested in asterisk over network mobility environment. For example
: when somebody is talking with
What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so
simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/
directory, asterisk loads 144 of them, omitting only chan_gtalk.so and
res_jabber.so.
Connected to Asterisk 1.4.1 currently running on monkey (pid =
We are running Alvarion VL 4.0. We know it doesn't work well over BA or
900. However, with 60 customers, all we error rates below 10% VoIP is
aweful.
On 3/7/07, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote:
What Alvarion stuff are you using? Alvarion is the top rated wireless
manufactor
I have done per user context programming for asterisk in the recent
past.
I am a professional contract programmer but my rates are reasonable and
you will get code that works.
Please contact me if you would like my help on this.
-Race
Race Vanderdecken
Code Tyrant, Inc.
[EMAIL PROTECTED]
828
I am using the * auto-dial out feature but don't want to have to
specify
a channel (Zap/G2/) to connect to the extension.
Current file I use:
Channel: Zap/G2/12127778866 # I have to specify a specific
channel
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your
Pablo Almido wrote:
Hi
I have a analog phone connected to my Gateway Handytone and registered
to Asterisk 1.4 I have configured my HandyTone 488
(in the section FXS Port) for make and receive calls, however I can
not transfer a call when it come via PSTN. But, when a call come from
via IP I
Hello,
I am trying to REGISTER asterisk to a SIP server, which is listening on Port
6060 (not 5060).
The sip.conf file contains
register=1847420:[EMAIL PROTECTED]:6060/1847420
maxexpirey=3600
defaultexpirey=120
But the REGISTER message is sent to Port 6060, but the
[EMAIL PROTECTED] asterisk]# cat res_mysql.conf
;
; Sample configuration for res_config_mysql.c
;
; The value of dbhost may be either a hostname or an IP address.
; If dbhost is commented out or the string localhost, a connection
; to the local host is assumed and dbsock is used instead of TCP/IP
== Parsing '/etc/asterisk/res_mysql.conf': [Mar 7 14:12:37] DEBUG[4380]:
config.c:844 config_text_file_load: Parsing /etc/asterisk/res_mysql.conf
Found
[Mar 7 14:12:37] WARNING[4380]: res_config_mysql.c:555 parse_config: MySQL
RealTime: No database host found, using localhost via socket.
[Mar
I think that is already set. Here is my queue.conf
[general]
persistentmembers = yes
autofill = yes
monitor-type = MixMonitor
[support]
musicclass = default
strategy = fewestcalls
timeout = 10
retry = 5
autofill=yes
autopause=yes
setinterfacevar=no
announce-frequency = 90
Something must be setup wrong on your radios. Alavarion is pretty good
when it comes to tech support. I would contact the distributer that
you bought it from and ask for help.
On 3/7/07, Matt [EMAIL PROTECTED] wrote:
We are running Alvarion VL 4.0. We know it doesn't work well over BA or
900.
Mike Hammett wrote:
[Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL
RealTime: Successfully connected to database.
[Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
[Mar 7 14:12:37] NOTICE[4380]: config.c:1174
7 mar 2007 kl. 20.48 skrev Bala Neelakantan:
Hello,
I am trying to REGISTER asterisk to a SIP server, which is
listening on Port 6060 (not 5060).
The sip.conf file contains
register=1847420:[EMAIL PROTECTED]:6060/1847420
maxexpirey=3600
defaultexpirey=120
But the REGISTER
Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output
Call File
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652
Priority: 1
Oh, We've been in contact with them.They said we need to purchase WLP..
which we are in the process of doing.. but then we found some other
interesting info out.I'll post a follow up here in a few days if I
remember.
On 3/7/07, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote:
Something
WaitTime stands for how long to wait until the call is considered NO
ANSWERED
Who can pickup a phone in 2 seconds, if not a robot? Try switch values
between Retrytime and WaitTime.
[]'s
MM
-Original Message-
From: Hall, Eric M. [EMAIL PROTECTED]
To:
OK now I fell like a a$$... Thanks for that kick in the butt !!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Melcon
Moraes
Sent: Wednesday, March 07, 2007 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Looks like it's a bug
http://bugs.digium.com/view.php?id=9172nbn=3
I have update to Asterisk SVN-branch-1.4-r58243 and will test it tomorrow and
report back to the list.
Eric Hall
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavio Ruiz
Has anyone run into the problem of having your FXO ports show up as FXS?
I believe that my drivers are loading because I see the channels appear
under /dev/zap. I am using Fedora Core 6 with the latest Zaptel 1.4.0
drivers and a 8 port FXO Xorcom Astribank. All of my indicator lights on
the
On Wed, Mar 07, 2007 at 08:39:36PM -0500, Lutgring, Sam wrote:
Has anyone run into the problem of having your FXO ports show up as FXS?
FXS channels have FXO signalling. Yeah, this is confusing.
An FXS port is the one to which you connect phones. FXO ports emulate
phones, and can connect to a
Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16
sip show channels
Always tends to show 100+ lines such as
192.168.1.241(None) 2e2872da-1d 00101/21507 unkn No
Rx: REGISTER
Never seem to go away
198 total peers on this server
All devices are behind NAT
Registration
I don't think this is a bug.
From UPGRADE.txt:
* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not
BJ
Here is the sip.conf file. Hints work great. The only problem is the queue is
sending calls to an agent that's on the phone.
[general]
rtcachefriends=yes
videosupport=yes
port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0 ; Address to bind to (all addresses
Hi all,
I've just been told from an ex workmate that my VSP (who I used to work
for) has put an anti flooding limit of 80 SIP messages per IP per minute
in place.
I run the phone system for a facility that has a lot of extensions, but
would rarely have more than 4 or 5 simultaneous
Um, is it possible to patch 1.2.4? We have some pretty busy
production systems and are not exactly excited about having to upgrade
from this version.
Is there no other way to protect our systems from this hole?
On 3/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Michelle Dupuis wrote:
Can
We run 1.2.4, do we have to upgrade? Is there a patch for this version?
At this point we REALLY don't want to upgrade and potentially
introduce a bunch of new issues and problems including our AGI's all
breaking due to SET and SETVAR etc..
Thanks in advance for any insight on this issue.
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--
Message: 17
Date: Wed, 7 Mar 2007 11:17:07 +0100
From: Thomas Deillon [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk 1.4.1 - Calling
/20070307/62a82c69/attachment-0001.htm
--
Message: 17
Date: Wed, 7 Mar 2007 11:17:07 +0100
From: Thomas Deillon [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Does anyone have a good suggestion for a automated solution to record calls
on certain interfaces and easily archiving them in a way which is easily
matched against CDRs? Also can someone suggest the appropriate protocol to
archive the recording when the conversations are transpiring in ulaw.
8 mar 2007 kl. 04.16 skrev Bill Gibbs:
Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16
“sip show channels”
Always tends to show 100+ lines such as
192.168.1.241(None) 2e2872da-1d 00101/21507 unkn
No Rx: REGISTER
Never seem to go away
198 total peers on this server
8 mar 2007 kl. 05.33 skrev voiplist:
Um, is it possible to patch 1.2.4? We have some pretty busy
production systems and are not exactly excited about having to upgrade
from this version.
Is there no other way to protect our systems from this hole?
You can apply the patch that was applied
https://sourceforge.net/projects/ccmanager/
please note that it is a beta version, i'd like to improve it but i'm
busy with work and university.
take a look and let me know.
nik
On 3/6/07, nik600 [EMAIL PROTECTED] wrote:
i've submittet the project to SF.
I have to wait 2 business days for
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