On Thu, Jul 9, 2009 at 4:41 PM, Miguel Molinammol...@millenium.com.co wrote:
Christian Gansberger escribió:
On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molinammol...@millenium.com.co
wrote:
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
--
Tzafrir
How about:
Exten = _X.,3,Dial(SIP/${ext...@carrier,60,M(fax-out))
[macro-fax-out]
exten = s,1,Set(FAXFILE=/root/test.tif)
exten = s,2,Set(LOCALHEADERINFO=WHO CARES WHO I AM ?)
exten = s,3,Set(LOCALSTATIONID=1-800-Who-CARES)
exten = s,4,SendFax(${FAXFILE})
- Original Message -
From:
Hi,
This week Tony Stankus, North American product manager of the Gigaset
line is our guest on VoIP Users Conference. I have had a two handset
S675IP in our small business for about a year now and my wife and I
both like the phone. But as a geek, I like it a lot more than she does
:)
6 SIP lines
(thank you gmail) so if you have DID without voice mail service, your
local Gigaset will handle the SIP channel as if it were
a PSTN line. This feature is selectable on a per account basis.
The phones also do g722 so they work with our ZipDX wideband bridge.
If you are considering new DECT
Hi
I am in Canada and I finally went a head with http://www.cigear.com. It was
quick painless and I did not have to deal with the customs and duties...
Tim
On Tue, Jul 7, 2009 at 5:54 AM, Tom O'Connor t...@twinhelix.org wrote:
On Mon, Jul 6, 2009 at 5:03 PM, Tony Mountifield
2009/7/9 Nick Hill t...@nickhill.co.uk
Hello Sasa
Carlos indicates that USB support may be available in chan_mobile but I
can't
find any references to it, and I think Oliver is looking for more info as
well.
That's perfectly true : I'm looking for more info for USB connectivity as
Hi,
Is there something available to add AEL2 syntax highlighting support to Kate
?
Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
I need to test the following scenario:
+---+ +---+
| asterisk 1| | asterisk 2|
+---+ +---+
| |
| |
___|__|___
| |
|
I have a basic config for AEL syntax highlighting for Kate if you would
like it.
- Brad
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, July 10, 2009 8:46 AM
the “sip show peers command returns
Name/username HostDyn Nat ACL Port Status
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
I ran grep on the sip.conf and it didn't find any IPs. Where would I add my IP?
I am guessing that the
Here is my physical network.
We have a Adtran router that is plugged into the Asterisk server and into the
circuit provided by my tel co.
the other nic in the Asterisk box is plugged into your lan switch
the phones are plugged into the lan switch
I can ping the phones from the Asterisk
Hi,
I have 2 asterisk servers link via IAX. and i'm trying to do a video call.
if 2 sip users are registered on the same server, the video works fine.
but if 1 sip user is on server 1 and sip user 2 on sip server 2. there's
no video at all. is it because call from sip server 1 goes to sip
Let's draw this out and let you fill in the blanks. Your asterisk server
has a name of foobar.com and an ip address of 192.168.23.1. phone 1 has ip
address of 192.168.23.2. phone 2 has ip address of 192.168.23.3.
Sip.conf should look this
[phone1]
type=peer
context=phones
Who is the carrier? What flavor of Asterisk are you using?
Regardless, the phones should register and be able to call each other and
other Asterisk apps if you have them in the dialplan.
If you go to the Asterisk CLI and turn on SIP debugging, do you get anything
at all?
also, change
Asterisk registers with the phones?
Obviously I have zero experience with these sets, but that is a new one.
Thanks,
Steve
On Fri, Jul 10, 2009 at 9:40 AM, Danny Nicholas da...@debsinc.com wrote:
Let’s draw this out and let you fill in the blanks. Your asterisk server
has a name of
To clarify, you have users 100, 101, and 102 on server 1 and 200, 201, and
202 on server 2. 100 can VC 101 and 102, but not 200-202. 100 can make a
voice call to 200-202. Have you checked your iax.conf to make sure all
codecs are functional?
-Original Message-
From:
My bad. Asterisk does not register with the phone. It can send out SIP
headers to make the phones re-register.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, July 10, 2009 8:45 AM
To: Asterisk
Sasa Bobek wrote:
Could not agree more. I had chan_mobile up and running with an older
version of Trix, but never managed to recreate it with the latest
versions. Other people I talked to even suggested that it was made on
purpose. With elastix the only problem I had was the missing
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else seeing this? My mail server logs don't show any issues.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
Doug Lytle wrote:
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else seeing this? My mail server logs don't show any issues.
Doug
I just did yes,
Don't know why :)
Dunc
On 10 Jul 2009, at 15:30, Doug Lytle wrote:
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else seeing this? My mail server logs don't show any issues.
Yeap. Happens quite regularly..
Anybody else seeing this? My mail server logs don't show any issues.
Digging a little further shows that ASSP blocked several 'Pharmacy'
spams from the list.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither
Dunc wrote:
Doug Lytle wrote:
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else seeing this? My mail server logs don't show any issues.
Doug
I just did yes,
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
hi sir
yes you're correct, voice call works from 100 to 200-202 but not video call.
on my iax i simply added:
videosupport=yes
allow=h264
allow=h263
TIA
Ron
Danny Nicholas wrote:
To clarify, you have users 100, 101, and 102 on server 1 and 200, 201, and
202 on server 2. 100 can VC 101
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Todd
Sent: Thursday, July 09, 2009 10:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Educational institutions: Your
Asteriskexperiences
Carrier is bandwidth.com
we are running Asterisk 1.6.1.1
i ran sip set debug on from the CLI
Once i did a module reload it started displaying all the debuging info. Here is
some of the debug info
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog
Extension 500 is registered just fine. 200 OK
Maybe you should start with a GUI version of Asterisk.
Try calling out via bandwidth with SIP verbose on and post your results.
Call the other phone and post verbose.
You do have logic in extensions.conf do you not?
Thanks,
Steve Totaro
On Fri,
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
constantly but not over the last few days. one for each list.
On Fri, Jul 10, 2009 at 10:30 AM, Doug Lytle supp...@drdos.info wrote:
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else
Hi There
I have an extension which is in a different country and is constantly
lagged (about 800ms). When anyone tries to call this extension we get a
No route to destination message.
Now I would have thought that the server should be able to find a route
to the destination seeing as the peer
so i filled in the info and now i get this when i run sip show peers
Name/username HostDyn Nat ACL Port Status
500/500127.0.0.1D 5060 OK (1 ms)
501/501127.0.0.1D 5060 OK (1 ms)
2 sip
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten =
I have the GUI setup and I setup users in the gui before. I still couldn't get
it to work. I don't have any SIP trunks setup via the GUI because I can't
figure out my settings and I was told I didn't need it to test extensions.
I am not sure what you mean by
Try calling out via bandwidth
You are running asterisk as a local service (127.0.0.1 is localhost). You
need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr).
This will make asterisk where your phones can talk to it and register.
_
From: asterisk-users-boun...@lists.digium.com
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve
Hi All
I am getting a strange prblem while installing TE420 on a Dell Poweredge T100
machine. I get a TE4XXX: version Synchronisation error nad the machine hangs
which needs a hard reboot . Its a new machine and if i install TE410P then
installation is successful. Strangely enough i have a
I saw 127.0.0.2, never seen that before. Loopback that I have seen is
127.0.0.1.
I always just bind to 0.0.0.0 since I have never really seen a point to
binding to a specific IP. I guess if you are dual homed and don't want
remote phones to work, but then you could just block that stuff in
Hello Everyone.
We have:
Asterisk 1.4.21.2
zaptel-1.4.11
libpri-1.4.5
Sangoma A101D Connected to a PRI
Cicso 7960G phones (About 30 of them)
We have a problem with dropped calls that has going on for a long
time. We get up to 5 dropped calls on a bad day. They all seem to be
incoming
Great i changed it to my ip here is the debug and sip show peers. phones still
say no service i get a dial tone when i pick it up and a busy signal when i
call the other extension.
Name/username HostDyn Nat ACL Port Status
500/50010.0.0.52
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at
Asterisk GUI-version : SVN-branch-2.0-r4962
Date: Fri, 10 Jul 2009 11:57:38 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] setting up phones
I saw 127.0.0.2, never seen that before. Loopback that I have seen is
127.0.0.1.
I always
Change the address in sip.conf, not the phone.
On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose sixfourimp...@hotmail.comwrote:
Great i changed it to my ip here is the debug and sip show peers. phones
still say no service i get a dial tone when i pick it up and a busy signal
when i call the other
We had the same problem using a Digium T1 card. We switched the coding to from
NI2 to 5ess and we haven't dropped a call since.
You will have to check with your service provider to see if they do 5ess.
Connor Spiess
Network Specialist
-Original Message-
From: Mark Engelhardt
Phone 1 has 500 in all of it's id's and connects to server 10.0.0.52?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Friday, July 10, 2009 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re:
yes
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 10 Jul 2009 11:20:29 -0500
Subject: Re: [asterisk-users] setting up phones
Phone 1 has 500 in all of it’s id’s and
connects to server 10.0.0.52?
From:
asterisk-users-boun...@lists.digium.com
Here is my conf files.
sip.conf
[general]
context=default
port=5060 ; UDP port for Asterisk
bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three
different IPs) 0.0.0.0 means any IP
srvlookup=yes ; Enable DNS SRV server
[500]
type=peer
context=phones
host=dynamic
Bind to 0.0.0.0
put your phones on DHCP if they are not already and reboot.
reload asterisk.
turn on sip debugging
call 501 from 500
post debug info.
i bet it rings.
On Fri, Jul 10, 2009 at 12:38 PM, Ott Rose sixfourimp...@hotmail.comwrote:
Here is my conf files.
sip.conf
[general]
Debug info is going to help the most here. Nobody is really going to look
at your configs.
I would also turn off lookup because if DNS fails, Asterisk doesn't care for
it much.
Try to hard code your IPs.
Thanks,
Steve
On Fri, Jul 10, 2009 at 12:42 PM, Steve Totaro
This is an age old Asterisk (and general telephony) problem. I can't blame
it all on Asterisk.
Never thought of the 5ess, filed in my memory bank as this is an age old
problem.
Too bad it happens with SIP providers and not just the little guys but XO
for instance.
I hear crackling. Cell
Conner,
I contacted my telco and they report they have a:
EWSD Siemens Central Office
Which does not support 5ess
Any other way around this? How did you determine changing to 5ess
would fix your problem?
Mark
On Jul 10, 2009, at 12:15 PM, Connor Spiess wrote:
We had the same problem
On Fri, 10 Jul 2009, Jared Mauch wrote:
I need the 'talking' information to better identify rogue people
on bridges.
I'm a 1.2 Luddite so I don't have all these fancy new features :)
A different solution to a similar problem.
I had problems with abusive callers in my conferences. I whipped
Steve,
Thanks for your thoughts. I am tearing out my last bit of hair on this
one.
We only use sip on our internal network to talk to the 7960s
We are getting drops from no-cell phone hard wired phones too.
Unfortunately There are too many drops for me to let this go. :(
Mark
On Jul 10,
DSS?
Ask them what other signaling they can support.
I would escalate every day if I were you. It is the only way to get things
done. Get it to the top and be mad, even if you are not.
When fixed, PRAISE everyone from top to bottom.
A level one tech will say Ah you are using Asterisk, we
How do I turn off the beeps in the head sets when customers are waiting
in the Queue?
Darryl Williams
Information Technology Manager
Direct: (214) 231-7325
Cell: (469) 583-6992
Fax: (262) 953-1929
Email: dwilli...@adeprocessing.com
Darryl Williams schrieb:
How do I turn off the beeps in the head sets when customers are waiting
in the Queue?
ringinuse=no in queues.conf and/or disable call waiting I guess.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan
I don't see my extensions in my extensions.conf file. I see a bunch of other
stuff but nothing that looks like this
exten = 500,500,Dial (SIP/500,20,tr)
I am guessing there should be something in there.
Date: Fri, 10 Jul 2009 12:44:56 -0400
From: stot...@totarotechnologies.com
To:
More info on my dropped call issue:
Here is a report on a dropped call from today:
Call Started echoing then cut out
Stats From the 7960 Stats Screen:
RxCnt:011853
TxCnt:010204
MaxJtr: 762
RxLost:
So, now I am starting to suspect that I have this problem:
Darryl Williams escribió:
How do I turn off the beeps in the head sets when customers are
waiting in the Queue?
Look for the option announce in queues.conf.
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
___
-- Bandwidth and
I know I'm doing something simple and wrong, but I can't quite figure it
out:
Example (executing system command): Action: Originate
Channel: Local/1...@dummy
Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System
Data: /path/to/script
I keep getting a Unable to request channel
For now, you need these two lines in your dialplan
- exten = 500,1,Dial(SIP/500,20,m)
- exten = 501,1,Dial(SIP/501,20,m)
This should let you dial your 2 extensions and hear MOH until it picks up
_
From: asterisk-users-boun...@lists.digium.com
Why not just Local/1 (unless your server is actually named dummy)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. G.
Sent: Friday, July 10, 2009 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
J. G. schrieb:
I know I'm doing something simple and wrong, but I can't quite figure it
out:
Example (executing system command): Action: Originate
Channel: Local/1...@dummy
Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System
Data: /path/to/script
I keep getting a
I don't use Asterisk-GUI but the general idea of a GUI is so you don't have
to modify files by hand. You use the graphical user interface to generate
the entries you need.
If you are using a GUI then don't touch the files.
Just download EVB (Easy Vox Box) and use the GUI.
If you want to mess
added that and still doesn't work. Is there a setting that could be set that
requires me to dial a # * or something before the extension number? Plus the
phones say no service? Should I reset them to factory and see if they pick up
the right extensions from Asterisk?
From: da...@debsinc.com
J. G. escribió:
I know I'm doing something simple and wrong, but I can't quite figure
it out:
Example (executing system command):
Action: Originate
Channel: Local/1...@dummy
Application: System
http://www.voip-info.org/wiki/view/Asterisk+cmd+System
Data: /path/to/script
I keep
I don't think the GUI is editing the conf files correctly. I am not sure I
have configure things right. At this point i think i am going to start from
scratch.
Date: Fri, 10 Jul 2009 16:19:52 -0400
From: stot...@totarotechnologies.com
To: asterisk-users@lists.digium.com
Subject: Re:
On Fri, 10 Jul 2009, Ott Rose wrote:
I don't think the GUI is editing the conf files correctly. I am not sure
I have configure things right. At this point i think i am going to start
from scratch.
Yea!
--
Thanks in advance,
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed both
On Fri, Jul 10, 2009 at 7:06 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Jul 10, 2009 at 6:44 PM, Wayne wa...@planetwayne.com wrote:
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi
On Fri, Jul 10, 2009 at 6:44 PM, Wayne wa...@planetwayne.com wrote:
Sorry to bump my own message - but had a mail server problem so don't
know if I missed any replys :(
Ta
Wayne.
Wayne wrote:
Hi all,
I've just built a new installation of CentOS release 5.3 (Final) and
have installed
Steve Totaro wrote:
If you are set on beta then read no further then the next line.
File a bug report with a core dump.
OK opinion time.
Your server is more than adequate.
For my tastes, you are beyond bleeding edge on the Asterisk front.
Simply my opinion but
On Fri, Jul 10, 2009 at 7:33 PM, Wayne wa...@planetwayne.com wrote:
Steve Totaro wrote:
If you are set on beta then read no further then the next line.
File a bug report with a core dump.
OK opinion time.
Your server is more than adequate.
For my tastes,
For a while now I've been looking for a good web based soft phone solution,
but so far no luck. A few solutions which I've tried, both Java based and
Flash based, either don't work, or had bad sound quality. I need something
which I could put on my productions server for my clients.
Seems like
On Fri, Jul 10, 2009 at 4:33 PM, Wayne wa...@planetwayne.com wrote:
Hi Steve,
Thanks for the pointers. I must admit - I was leaning towards 1.6 as
this apparently has support for SIP over TCP (?). My end goal with this
was to try and get Asterisk talking to Exchange 2007 servers unified
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