Hi All;
Any one can advise for a good stable open source video
conference or video server?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is 193.111.194.111
IP Phone (A): 192.168.0.1 and its default gateway is:
Hi All;
Is there an IP Phones working with Asterisk that come
built in with VPN Client? And what the VPN server it
works with it fine?
Regards
Bilal
Never miss a thing. Make Yahoo your home page.
Hello List,
Im just dipping my feet into the asterisk world, and im having major
fxo problems
Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons
(from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a
Debian Etch box, with 1gb ram, running all of the services for my
On Mon, Dec 10, 2007 at 10:06:02AM -0500, Peter Pauly wrote:
Does anyone know how to customize the order of the soft keys on a 7960
running SIP? All the documentation I could find is CallManager
related. Specifically, I want to move the transfer function to the
first set of buttons during a
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Dear all,
I've a live system that needs to be upgraded but, before I proceed to the
upgrade I want to assure the rollback process.
That's why I'm requesting your feedback, in fact this asterisk in live
system isn't going so bad but the upgrade is essential
NOTICE that the upgrade will keep
Snom
2007/12/11, bilal ghayyad [EMAIL PROTECTED]:
Hi All;
Is there an IP Phones working with Asterisk that come
built in with VPN Client? And what the VPN server it
works with it fine?
Regards
Bilal
Benjamin Jacob wrote:
Hello ppl,
Am totaly new to this zap thingy.. zapped I would say I am! (couldn't
resist that cliche...).
Just like sipp for testing SIP channels, do we have any such tools to
test zap channels?
You can try PBX Testing Framework. It's using SIP by default, but as
I have a working system with two fxo and two fxs channels. I recenlty got an
IAX2 account I would like to use also.
While I have gotten the IAX2 channel to register, it remains non functional,
as the incoming calls, go nowhere and the outgoing calls attempt to go out over
the ZAP channel. I
Hi :)
sandeep.s wrote:
my sip phone is unreachable for external network(global ip)
Some of my SIP phones are unreachable as well. That's because
I unplugged the power cord.
You need to provide a bit more information or else the solution
may be described like make sure everything is set up
]:4] AGI(SIP/2290-09b18a68,
recordingcheck|20071211-100351|1197374631.0) in new stack
[Dec 11 10:03:51] VERBOSE[12935] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
[Dec 11 10:03:51] VERBOSE[12935] logger.c:
recordingcheck|20071211-100351|1197374631.0: Outbound
Greetings, List.
I'm having a problem where my recorded calls are skipping every 4-5
seconds are so. I can hear the caller (or callee) just fine and then a
second or so of silence followed by the person talking again. I'm
saving my calls as .gsm files and it's worked fine for the past 11
I have 800 kbps in both directions reliably at the endpoint location. When I
was testing, there weren't any computers in the office, or any other phones.
The server has a 10 Mb ethernet connection in a datacenter, and I usually
don't see more than 8 channels at once, so I don't think it's
On Tue, 11 Dec 2007 11:58:06 +, Robert Lister wrote:
Every so often I think that there must be a better handset out there, and
indeed there are better handsets out there that allow things like call
reject, Busy lamp field etc. (SIP feature-wise the Cisco phones are very
basic.) but the
Hi Kevin -
I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC
($1610).
Would we really suffer
without the onboard echo cancellation?
Each situation is different, but I have a client that had significant
problems with echo on their PRI. Asterisk's software EC (any of
Hi Alex -
Is there a way to dynamically alter the sip.conf properties of a SIP peer
in runtime without doing a SIP reload?
realtime (i.e. database)?
- Noah
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
I don't know of a way without reloading. Realtime still needs a sip
reload.
Look at the dial command. There are options that you can add that will
disable re-invites per call.
Doug Gillespie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Hi -
I guess that's my question. Is this the standard method of doing faxing?
Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
I've never used a TDM880 for this purpose, but I've used multiple
TDM400's in this capacity (PRI - TE4XXP - TDM400 - Fax Machine),
and it works
When making or receiving a SIP call via my service provider, I get the
following message logged by Asterisk:
Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if possible.
Client IP: xxx.xxx.xxx.xxx
Since
Hi,
How can I merge 2 gsm files into a single file? I have tried to use
soxmix as below but failed.
soxmix 1.gsm 2.gsm 1-2.gsm
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or
Roger,
The seize ack timeout problem is because libmfcr2 is expecting a
response ( an ACK ) from the far end and it does not arrive in a R2
variant dependant amount of time. Which protocolvariant do you have
configured in unicall.conf?
This is how the process to start a call goes:
1. When you
Realtime only needs a sip reload if you are using static realtime, if
you use the sippeers realtime it works just fine. See
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
Note that the settings change will only take effect when your client
re-registers, so you may want to set a
On Tue, Dec 11, 2007 at 11:26:21PM +0800, Rilawich Ango wrote:
Hi,
How can I merge 2 gsm files into a single file? I have tried to use
soxmix as below but failed.
soxmix 1.gsm 2.gsm 1-2.gsm
The GNU coreutils are shipped with a special[1] tool for this task:
cat 1.gsm 2.gsm 1-2.gsm
[1]
Happens.
On Tue, 11 Dec 2007, sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Sounds like good security practice to me. YMMV.
CP
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
On Dec 10, 2007, at 7:45 AM, Michael Melia Jr. wrote:
I haven't found outcall that confusing though I do agree that a TAPI
Driver that makes use of the available outlook call functions will
make
for the easiest, most streamlined user experience.
I also agree that these convenience and
Hi
i need to install a server with this hardware:
1 OpenVox B800P
1 OpenVox A800P01
4 OpenVox FXS-100 FXS100
4 OctWare SoftEchoSOFTECHO
Do you suggest 1.2 or 1.4 branch?
Is now 1.4 stable ?
I've tried 1.4 the last year but i've experienced many
Moises,
Thank you for your reply and the lesson of MFC/R2 !
My configs for the unicall.conf is:
[channels]
language=br
context=from-pstn
usecallerid=yes
hidecallerid=no
immediate=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
On Tue, 2007-12-11 at 00:14 -0800, bilal ghayyad wrote:
Hi All;
My Asterisk has a public IP address, how can we let
two IP Phones in different site and both are behind
NAT (each one has a private IP address) to call each
other?
In other words,
Assuming Asterisk IP Address is
Hi,
One of Asterisk features is fax2mail.
As a good share of incoming faxes can be considered as advertising spam,
does it make sense to use email anti-spam features to filter them ?
I can't foresee any practical way to do so but I would be very curious to
discuss about it.
Regards
That's exactly the information I'm looking for, thanks Noah.
On Dec 11, 2007 10:16 AM, Noah Miller [EMAIL PROTECTED] wrote:
Hi -
I guess that's my question. Is this the standard method of doing faxing?
Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
I've never
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 12/11/2007 11:23:29 AM
Hi,
One of Asterisk features is fax2mail.
As a good share of incoming faxes can be considered as advertising spam,
does it
On Tue, Dec 11, 2007 at 06:23:29PM +0100, Olivier wrote:
Hi,
One of Asterisk features is fax2mail.
As a good share of incoming faxes can be considered as advertising spam,
does it make sense to use email anti-spam features to filter them ?
I can't foresee any practical way to do so but I
Tony Mountifield wrote:
When making or receiving a SIP call via my service provider, I get the
following message logged by Asterisk:
Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible.
Hello
I'm looking at my options to build a compact, silent, headless
Asterisk server to handle one or two FXO ports. Out of curiosity, I
got one of those babies on eBay for 20E:
http://silicon-verl.de/home/flo/software/netstation-8364/
Before I spend time on this, can someone tell me
Hi Oliver;
Thanks alot for your reply.
What the needed VPN server? Cisco, Juniper, Planet ?
Does it use IPSec or PPTP?
Regards
Bilal
-
Snom
2007/12/11, bilal ghayyad [EMAIL PROTECTED]:
Hi All;
Is there an IP Phones working with Asterisk that
come
built in with VPN Client?
Does anyone know how I could integrate my Asterisk setup with Outlook so
that when I click on a phone number is my outlook address book it will
dial the number and ring my SIP phone so that I can just pick it up? I
am interested in this integration for WinXP with Outlook 2003 and
I will be out of the office from Dec. 10-13.
Thanks,
Dan Baker
Ursuline Academy
314-984-2828
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Tue, 11 Dec 2007 10:09:34 +, Chris Boczko
[EMAIL PROTECTED] wrote:
Im just dipping my feet into the asterisk world, and im having major
fxo problems
I'm no Asterisk expert.
The X100p is a cheap clone i got off ebay for a tenner, so im not
expecting much, i know they have echo issues, but
Chris Boczko wrote:
Hello List,
Im just dipping my feet into the asterisk world, and im having major
fxo problems
Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons
(from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a
Debian Etch box, with 1gb ram, running
Anyone,
could you please suggest the latest stable release for asterisk.
-Jai
On Dec 10, 2007 9:08 PM, Jai Rangi [EMAIL PROTECTED] wrote:
I am planning to upgrade my asterisk to
Asterisk
We're trying to get a SIP peer going between our asterisk box and our
provider. It should then ring our phone.
The call does come in and it does execute the extension in the dial
plan. But the provider says they never get a 200 OK back and therefore
they send another INVITE and then after a few
The snom 370 used a OpenVPN client.
See http://en.wikipedia.org/wiki/OpenVPN and
http://wiki.snom.com/Networking/VPN (that link contains a slash, but is also
linked on http://wiki.snom.com/Main_Page).
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im
Hello List,
Thanks for the replies...currently
Still doing the same after adding channel=1 to the /etc/zaptel.conf
my zapata.conf looks like
orange:~# more /etc/asterisk/zapata.conf
[trunkgroups]
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
Are there any phones whose display can show queue statistics, ie:
calls waiting, etc, on the phone itself without too much trouble with
Asterisk? Especially while the phone is in use (on a call)?
___
--Bandwidth and Colocation Provided by
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When making or receiving a SIP call via my service provider, I get the
following message logged by Asterisk:
Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise
support
The old x100p cards where 5 volt pci cards. I had this same problem and it was
the type of pci slot that I had the card plugged into.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Boczko
Sent: Tuesday, December 11, 2007 1:02 PM
To: Asterisk
Hi * users,
Is there any way to check if a SIP user is currently available or not
from the dialplan? By available I mean if a SIP call can be made to
that user.
Thanks
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
On Tue, 11 Dec 2007 11:05:24 -0600, Carlos Chavez
[EMAIL PROTECTED] wrote:
The only thing you need to do is set nat=yes when you configure the
phones in Asterisk. You may need to use a STUN server in case the
phones do not properly see the outside address. Once the phones
register they
On Tue, Dec 11, 2007 at 10:09:34AM +, Chris Boczko wrote:
Hello List,
Im just dipping my feet into the asterisk world, and im having major
fxo problems
Im running Asterisk (from svn) + libpri (from svn) +
You don't really need libpri, as you have a simple analog card. Actually
it
You can use sip show peers. If an IP address is shown then the user will be
available.
Henk
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
See Inline
On Tuesday December 11 2007 2:20 pm, Jonn R Taylor wrote:
The old x100p cards where 5 volt pci cards. I had this same problem and it
was the type of pci slot that I had the card plugged into.
Jonn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The ChanIsAvail() application provides a rudimentary means of doing this,
but it is not reliable in that it does not cover certain categories of
channel state that might be considered subjectively untenable for call
placement, i.e. a user is on the phone and you do not want to send them
a call by
Roger,
You can try to pass the protocolvariant like this:
protocolvariant=br,20,4,x,max-seize-wait-ack=3000
This deserves a little bit of more explanation.
br = Brazil
20 = ANI digits
4 = DNIS digits
x = this is just a hack to be able to work with defaults and specify
the next value.
Que has comio?
Que has comio?
(Its a joke in spanish-chanante)
2007/12/11, CunningPike [EMAIL PROTECTED]:
Sounds like good security practice to me. YMMV.
CP
sandeep.s wrote:
Hi,
my sip phone is unreachable for external network(global ip)
Thanks,
sandeep.s
Are the cordless phones on the 480i CT from Aastra registered independently in
Asterisk? Such that if I have 5 of the cordless phones hooked up, each one is
it's own extension?
This e-mail, facsimile, or letter and any files or attachments transmitted with
it
Nope they all share the same extensionif you want independent
wireless handsets and you don't want to use WiFi phone, you'd have to
look at Aastra's SIP DECT product, or Kirk, or something along those
lines.
Cory J Andrews
From: [EMAIL PROTECTED]
No. All lines/extensions are registered to the base phone and the
handsets access the lines via the base unit. You can have multiple
simultaneous calls.
Jeremy Mann wrote:
Are the cordless phones on the 480i CT from Aastra registered
independently in Asterisk? Such that if I have 5 of the
On Tue, Dec 11, 2007 at 03:25:05PM -0500, John Millican wrote:
See Inline
On Tuesday December 11 2007 2:20 pm, Jonn R Taylor wrote:
orange:~# more /etc/asterisk/zapata.conf
[trunkgroups]
[channels]
[snip]
channel=1
I believe this should be channel=1
Those two would have the same
Noah Miller wrote:
Each situation is different, but I have a client that had significant
problems with echo on their PRI. Asterisk's software EC (any of the
varieties) was not able to fully compensate for it.
When you say Asterisk's software EC do you mean the EC that comes with
Asterisk or
Hello,
I tried to install the asterisk 1.4.15 and I am not able to start it. I get
the segmentation fault error. What might be wrong, where I can look for a
clue.
Also could some one PLEASE suggest the most stable version of asterisk.
-Jai
___
We are looking for an Asterisk tech in the Brisbane area...ideas?
PaulH
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Wed, 12 Dec 2007, Paul Hales wrote:
We are looking for an Asterisk tech in the Brisbane area...ideas?
Try the -biz list.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote:
Hi
Does anyone have any recommendations of an SMS gateway which you can
just sign up for on a pay-as-you-go basis for testing, for use with
Asterisk?
Thanks
Robert McNaught
In and Out Bound SMS from *, or just * - SMS? If
Im looking to just test the concept of sending SMS texts from *.
When you say a provider? What kind of provider do you mean?
Robert
On Tue, 2007-12-11 at 17:44 -0600, Greg Oliver wrote:
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote:
Hi
Does anyone have any recommendations
In my experience the most stable asterisk is the one that runs and
runs and never crashes.
On 12/11/07, Jai Rangi [EMAIL PROTECTED] wrote:
Hello,
I tried to install the asterisk 1.4.15 and I am not able to start it. I get
the segmentation fault error. What might be wrong, where I can look for
nat
On 12/11/07, Rob Schall [EMAIL PROTECTED] wrote:
We're trying to get a SIP peer going between our asterisk box and our
provider. It should then ring our phone.
The call does come in and it does execute the extension in the dial
plan. But the provider says they never get a 200 OK back and
And that version name/number is ???
:)
-Jai
On Dec 11, 2007 4:17 PM, C F [EMAIL PROTECTED] wrote:
In my experience the most stable asterisk is the one that runs and
runs and never crashes.
On 12/11/07, Jai Rangi [EMAIL PROTECTED] wrote:
Hello,
I tried to install the asterisk 1.4.15 and
On Tue, 11 Dec 2007 18:51:44 +0100, Vincent wrote:
Hello
I'm looking at my options to build a compact, silent, headless
Asterisk server to handle one or two FXO ports. Out of curiosity, I
got one of those babies on eBay for 20E:
http://silicon-verl.de/home/flo/software/netstation-8364/
On Tue, 11 Dec 2007 19:24:52 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
There's no reason why that could not work for you. With a 266 MHz CPU
you have a platform roughly comparable to a Soekris Net4801. That means
limited transcoding.
Thanks for the tip on the HP T5700. There's one for sale
Hello Everyone,
We have recently installed a pair of Trixbox servers in for a client of our.
They have two locations, with one server each. The servers terminate 3 standard
POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers
(1GB Ram, Raid 1 160GB HDD, Dual Core Xenon
sorry, but that depends on much more than the version, while I have
never personally tried 1.4 the latest 1.2 works for me, however I have
never yet passed the stage of never as far as time goes, since
tomorrow is another day.
now to your problem, if you start asterisk with -cv at loading
what
On Wed, 12 Dec 2007 02:48:31 +0100, Vincent wrote:
On Tue, 11 Dec 2007 19:24:52 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
There's no reason why that could not work for you. With a 266 MHz CPU
you have a platform roughly comparable to a Soekris Net4801. That means
limited transcoding.
Thanks
Hi,
We had modified some configuration in our cisco 800 series router. We set
all the UDP packets from our servers to ip precedence 5 and also allocate 75%
of bandwidth for UDP packets.
However we still facing latency and low volume problem. Is it our 512k
outbound bandwidth not
How are the calls being transferred from Box A to Box B?
On what box is the receptionist registered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To:
Do an RTP analysis with Wireshark of a sample call. That could
probably narrow down the source of the problem. I would suspect you
will either see some jitter or packets out of order.
Daniel Cole wrote:
Hello Everyone,
We have recently installed a pair of Trixbox servers in for a client
The two boxes are labeled as per the town they are in: Leongatha and Korumburra.
The receptionist is in Korumburra.
When a call comes in off the PSTN in Leongatha, the first number in the call
queue is the receptionist. If she answers it, then the media flow looks like
this:
PSTN - Leongatha
What codec are you using?
PaulH
On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote:
Hello Everyone,
We have recently installed a pair of Trixbox servers in for a client
of our. They have two locations, with one server each. The servers
terminate 3 standard POTS lines into a Sangoma
G729 All Around.
Daniel Cole (CCNA)
Technical Support
[http://www.hugonet.com.au/clients/hugonet.gif]
Ph: 1800 424 683
Fax: 03 5221 7659
e: [EMAIL PROTECTED]mailto:[EMAIL PROTECTED]
w: hugonet.com.auhttp://www.hugonet.com.au/
Hi Paul,
Where abouts exactly is the best place to get these figures from?
I have been checking iax2 show netstats, which does give some figures. These
appear not to be accurate though, as when there are multiple inter-site calls,
the result for one channel of audio can show no jitter or
'iax2 show channels'maybe
I have a feeling this is going to be one of those ugly ones where it's
going to be a pain to troubleshoot...
Offhand - have you tested 'trunk=yes' vs 'trunk=no'?
PaulH
On Wed, 2007-12-12 at 17:00 +1100, Daniel Cole wrote:
Hi Paul,
Where abouts exactly
82 matches
Mail list logo
