We have 2 linked servers (IAX) and I would like our smartphone to be
able to show the line status of an extension on the linked server.
Meaning if an extension on the linked server is being used, I want the
light corresponding to that extension to be lit up on the smartphone
the same way it works
On 17:54, Mon 08 Oct 07, D4rk F1ber wrote:
So yes I am asking because I am unimaginative and need ideas on
selling this to the wife. :-) That and I am just curious about what
others feel are useful uses for it within the home, and what others
get excited about regarding it all.
What did the
D4rk F1ber wrote:
snip /
One of the next projects for me personally is to get a SIP client for
my Cingular/ATT 8525, it has wifi and hsdpa running Windows Mobile 6
and I am certain I have run across SIP clients before for these
things. Be fun to play with and get working.
So yes I am
On 10/9/07, D4rk F1ber [EMAIL PROTECTED] wrote:
So yes I am asking because I am unimaginative and need ideas on
selling this to the wife. :-) That and I am just curious about what
others feel are useful uses for it within the home, and what others
get excited about regarding it all.
I
Michiel van Baak wrote:
snip /
What did the trick for me is integrating it with MythTV.
When the phone rings my tv pauses, and starts recording on
the harddisk. Once the call is over my wife has 15 seconds
to go back to her seat before the tv resumes.
Way cool :-)
--
The way out is open!
Alan Lord wrote:
I run a small Open Source consulting/training company here in the Uk
and am starting to build an * server so that myself and my business
partner (who both work from our respective homes) are communicating
properly.
I have a couple of colleagues who also work from home -
Maybe I was lucky, but a client of mine has a 24 FXO TDM2400 and works
like a charm :)
l.
On Sun, 07 Oct 2007 03:06:52 +0200, C F [EMAIL PROTECTED] wrote:
Because they tried competing with the channel bank market. But guess
what, it has only one competitive edge, it's cheaper. But if you
I understand you - it's better to settle down for a few hours with a book
of the dead-tree kind. :)
You could also try SIP Beyond VoIP - it's not just on SIP, but it gives
you a broader usage/adoption scenario.
l.
On Mon, 08 Oct 2007 13:42:01 +0200, Justin Case
[EMAIL PROTECTED] wrote:
On Tue, 09 Oct 2007 01:05:41 +0200, Anselm Martin Hoffmeister
[EMAIL PROTECTED] wrote:
Asterisk can do all of that. Something along the lines of
Thanks a lot for the help :-)
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
I'm trying to find the right Digium card for the
Dell 2950
Dell 2850
HP DL380 G3
HP DL360 G3
Are these 3.3v or 5.0v machines ? I am out of the office, and need to
buy a card today.
I am looking at either the TE407 or TE412, and would appreciate any help. :)
Julian
On 10/9/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I'm trying to find the right Digium card for the
Dell 2950
Dell 2850
HP DL380 G3
HP DL360 G3
Are these 3.3v or 5.0v machines ? I am out of the office, and need to
buy a card today.
I am looking at either the TE407 or TE412, and
D4rk F1ber wrote:
So yes I am asking because I am unimaginative and need ideas on
selling this to the wife. :-) That and I am just curious about what
others feel are useful uses for it within the home, and what others
get excited about regarding it all.
I do trunks/terminations so its easy
If it hasn't already been done I am looking to put together a team to write
drivers for this DS3 card to interface asterisk.
http://www.imagestream.com/PCI_921-CDS.html
The card itself seems reasonable and I believe we can make it work. As soon
as I have positive feedback to begin the
Hello everybody,
is it possible that, when Asterisk is executing extensions reload, if I issue
another extensions reload I can mess up the dialplan?
If so, I think that the correct behaviour should be using a lock for the
dialplan and letting the second extensions reload wait for the first to
Before you put any work into this... ask yourself... what exactly are you
hoping to accomplish? There is no way one system can handle a DS3s worth
of traffic... therefore, what good would this do?
On 10/9/07, Tim King [EMAIL PROTECTED] wrote:
If it hasn't already been done I am looking to
Hi
if i want to use sip client to connect to my asterisk pbx do i need to
run a sip server ?
If so can you point me in the direction of a good howto for asterisk and sip ...
Thanks
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want a separate context
for each going into the PBX. This worked very well with IAX.
I want to use SIP
I think that reading an introductory book AND the rfc is the best
choice to learn sip. The rfc is very well written and is a more
complete reference. Wiresharking sip conversations could help you too.
On 10/9/07, Lenz [EMAIL PROTECTED] wrote:
I understand you - it's better to settle down for a
Hey,
This is weird, I wonder if anyone has an explanation? If I call a SIP
server and inject DTMF with a wait in it, my router will then lock up
causing asterisk to lose Internet connectivity obviously, but also
making it very hard to see what happens. It appears that if there are
no 'w' in the
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote:
Before you put any work into this... ask yourself... what exactly are
you hoping to accomplish?
I can imagine it be used as a TDM-SIP gateway but if I needed such a box
I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or
look at
A few SGI boxen with Numalink could probably handle it just fine.
Thanks,
Steve
Matt wrote:
Before you put any work into this... ask yourself... what exactly are
you hoping to accomplish? There is no way one system can handle a DS3s
worth of traffic... therefore, what good would this do?
It's not the Ethernet interface that would be the issue. The zaptel
framework wouldn't be able to handle it with the way it uses interrupts.
On 10/9/07, Patrick [EMAIL PROTECTED] wrote:
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote:
Before you put any work into this... ask yourself... what
You don't need to define a gatekeeper, it's optional.
It's not official documentation and not prove that, although I think you
could believe in it.
http://www.voip-info.org/wiki/view/Asterisk+oh323+channels
Regards
On 10/9/07, brahem mehdi [EMAIL PROTECTED] wrote:
thanks Machado,
but i have
Hi Philipp,
Thank you for your response to my question. I am working on a project
which uses Asterisk as the voice engine. I need to get the ingress
and egress sip call id for a call to write call CDR. (Asterisk CDR
does not meet our customer requirments). If there is no any easy way
On 10/9/07, Gregory Machin wrote:
Hi
if i want to use sip client to connect to my asterisk pbx do i need to
run a sip server ?
If so can you point me in the direction of a good howto for asterisk and sip
...
install any sip client on your workstation computer
and point it to your
I did not look at the specs of the card but if it has inboard DSPs, it
may work just fine in a high end box.
Thanks,
Steve
Matt wrote:
It's not the Ethernet interface that would be the issue. The zaptel
framework wouldn't be able to handle it with the way it uses interrupts.
On 10/9/07,
I had a friend yesterday showing me his new T-mobile blackberry with
WiFi Voip.I could not believe it until I actually saw him making
calls. There is no T-Mobile cell coverage at my house but he was able
to simply access the WiFi router and make the call. It appears this
VoIP offering
You can capture the sipcallid from the manager output. The cool part is
that the sipcallid is the same on both sides of a call. So,
AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as
AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
Ray
On Tuesday 09 October 2007 10:14:23 Matt wrote:
Before you put any work into this... ask yourself... what exactly are you
hoping to accomplish? There is no way one system can handle a DS3s worth
of traffic... therefore, what good would this do?
Whatever gave you the notion that a modern PC
its IMS
/b
On Oct 9, 2007, at 10:39 AM, Andres wrote:
I had a friend yesterday showing me his new T-mobile blackberry with
WiFi Voip.I could not believe it until I actually saw him making
calls. There is no T-Mobile cell coverage at my house but he was able
to simply access the WiFi
http://www.imagestream.com/PCI_921-CDS.html
This card can do it. I have spoke with them about it and its very
capable of doing what is needed for a DS3 in a standard linux box.
/b
On Oct 9, 2007, at 10:42 AM, Andrew Kohlsmith wrote:
On Tuesday 09 October 2007 10:14:23 Matt wrote:
Before
Patrick wrote:
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote:
Before you put any work into this... ask yourself... what exactly are
you hoping to accomplish?
I can imagine it be used as a TDM-SIP gateway but if I needed such a box
I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco
Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want a separate context
for each going into the PBX. This worked very well with
Andrew Kohlsmith wrote:
On Tuesday 09 October 2007 10:14:23 Matt wrote:
Before you put any work into this... ask yourself... what exactly are you
hoping to accomplish? There is no way one system can handle a DS3s worth
of traffic... therefore, what good would this do?
Whatever gave you
On 10/8/07, Alex Balashov [EMAIL PROTECTED] wrote:
Greetings,
I have a very basic equal-weight ring-all queue set up in queues.conf:
[sales-queue]
;music = default
strategy = ringall
periodic-announce-frequency = 20
announce-holdtime = no
timeout = 15
maxlen = 0
member = SIP/[EMAIL
Yep all the carriers are looking to offer 'voip' services sooner rather
than later. Basically it uses the wifi point to access the mobile
switching network.
Cool part is you will soon be answering your Verizon home phone on your
cell when you are 'within range' or your home network.
Regards,
This looks very promising. All eggs in one basket, but promising...
Any idea on price?
The PCI 921-CDS utilizes the Mindspeed CX28500 chipset to provide
support for the card's host PCI bus interface, which can burst data at
speeds up to 780 Mbps, or 390 Mbps full duplex. The CX28500 also
On 10/9/07, Brian West wrote:
http://www.imagestream.com/PCI_921-CDS.html
[...]
off-topic :
I saw Imagestream at the Ohio Linuxfest a weekend ago.
Also picked up a few literature bags by Digium :-)
--
___
--Bandwidth and Colocation
Will this work backwards? When I'm at home instead of my cell ringing
have the home phone ring? Why would anyone give up revenue from minutes?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Tuesday, October 09, 2007 12:03 PM
To: [EMAIL
Steve Totaro wrote:
Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want a separate context
for each going into the PBX. This
I have Asterisk 1.2.13 installed as a Debian package and I edit only
sip.conf and extensions.conf in this way:
sip.conf:
[general]
realm=work.com.ar ; Realm for digest
authentication
bindport=5060 ; UDP Port to bind to (SIP standard port
is
I have started the open source project to get this going. I am working
directly with the manufacture to form agreements and gain access to the
hardware and source code for their drivers. The list price for the card is
$4,995.00 USD. I will keep everyone posted and will have site for
development
Eric ManxPower Wieling wrote:
Steve Totaro wrote:
Steve Totaro wrote:
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it
worked fine except for audio issues that I believe are directly related
to IAX2 in version 1.2.x. I have four PRIs and want a separate context
for
This line gives you the clue:
Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial
argument takes format (technology/[device:]number1
Your dialplan should have Dial(SIP/user1) rather than Dial
(SIP,user1) / instead of ,
Give that a try.
--
Aubrey Wells
Senior
Still having no luck with this scenario. Has anyone else experienced
problems with em wink lines?
I'm thinking that there could be problems with the timing settings in
zapata.conf, but documentation is pretty light.
How could the telco not be receiving enough digits when it works for 500
On Tue, 2007-10-09 at 09:55 +0200, Michiel van Baak wrote:
On 17:54, Mon 08 Oct 07, D4rk F1ber wrote:
I am just curious about what
others feel are useful uses for it within the home, and what others
get excited about regarding it all.
What did the trick for me is integrating it with
I'm already doing that.
/b
On Oct 9, 2007, at 11:31 AM, Tim King wrote:
I have started the open source project to get this going. I am working
directly with the manufacture to form agreements and gain access to
the
hardware and source code for their drivers. The list price for the
card
A critical lesson I learned was not to rely to heavily on the AMI,
especially when there are other ways of doing the same thing that are
just as simple.
I suggest, rather than using AMI originate, mv or ftp .call files.
Thanks,
Steve
Whit Thiele wrote:
Still having no luck with this
Competition is a good thing. Let's say you fail or your implementation
is not as robust as the other project or visa versa. Just as long as
the hardware vendor is different, it should be a good thing. If it the
same hardware vendor, then maybe you two should work together.
Thanks,
Steve
Just to be clear, I would eliminate the AMI as the culprit first. I
have seen extensive use of the AMI cause all kinds of flaky behavior.
Zaptel, timing, or EM wink may be working perfectly but the AMI is
borking everything up, thats my thought anyways.
Thanks,
Steve Totaro
Steve Totaro
You apparently don't realize you're talking to. Thats ok, You keep
working on it from your angle. We are evaluating when the time is
right to implement this. We aren't doing this for Asterisk we are
doing it for FreeSWITCH.
/b
On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote:
Technically anything is possible - a few years ago I was working with
Siemens to implement something called Openscape which never took off in
the USA but basically was a web based application which allowed company
users to redirect their office phone numbers from the web to their
mobile or home
Hello Don,
thanks for the helpful pointers, i'll push my quotes on these and
hopefully they will be accepeted.
The only drawback on this is the fact that i would have to use an ATA
to complete the loop. This will rais the unit cost of the deployment.
I was thinking of usin SOEKRIS installed with
Don't take it personally. I have been on this list about as long as
you. BKW (Next!) Ego can be good but let's not become egomaniacs
shall we?
I am not working on it from any angle, and would probably never
*entertain* using such a device.
I prefer tried and true DS3 MUXs such as the
BTW, this is the wrong list if it not for Asterisk. It has absolutely
nothing to do with Asterisk.
Please post to the appropriate FreeSwitch list.
Thanks again,
Steve Totaro
Brian West wrote:
You apparently don't realize you're talking to. Thats ok, You keep
working on it from your
Well we are plugging it in the OpenZAP abstraction layer we have
already started on. This is usable by Asterisk also so asterisk
would benefit from it.
http://fisheye.freeswitch.org/browse/OpenZAP
/b
On Oct 9, 2007, at 12:31 PM, Steve Totaro wrote:
BTW, this is the wrong list if it not
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, Brian West [EMAIL PROTECTED] wrote:
You apparently don't realize you're talking to. Thats ok, You keep
working on it from your angle. We are evaluating when the time is right to
implement this. We aren't doing this for
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
What about Multi-NAT where a dedicated public IP is mapped to the
private IP of the asterisk box..
ie UA - NAT - Internet - Multi-NAT - Asterisk
http://www.draytek.co.uk/support/kb_vigor_multinat.html
And what was the purpose of this?
/b
On Oct 9, 2007, at 1:32 PM, Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, Brian West [EMAIL PROTECTED] wrote:
You apparently don't realize you're talking to. Thats ok, You
keep working on it from your angle. We are
Yes, I knew who I was talking to and now I know a little more about you
Matt, that was totally uncalled for.
Thanks,
Steve Totaro
Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, *Brian West* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You
Matt,
I talk very openly about this issue. It was very rude of you to
bring this up. This plea was total bullshit. If you want to know
the whole story feel free to call me and talk about it.
918-424-9378... anyone can call me and ask me questions about it.
The plea was a deal worked
I have tried it with the best result of one way audio after spending a
few days doing everything imaginable. This is the only scenario where I
suggest using IAX.
Thanks,
Steve Totaro
WipeOut wrote:
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
On 10/9/07, Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
Hey,
I am not sure what your point is, are you trying to shame
West on this list with your post ?
He is a contributor to the asterisk movement, which is the
purpose of these lists.
This was uncalled for.
Perhaps it was uncalled for. However, if I were to consider using
FreeSwitch I would want to know who was/is behind it.
On 10/9/07, Brian West [EMAIL PROTECTED] wrote:
And what was the purpose of this?
/b
On Oct 9, 2007, at 1:32 PM, Matt wrote:
On 10/9/07, Matt [EMAIL PROTECTED] wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
Fascinating. Not really. Anyway, how is this related to Asterisk?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Our office does not have a PA system, and in our current phone system we have a
certain extension that we dial that pages over the speaker of all the phones in
the office. Does Asterisk support this feature? If so, could someone tell me
the best way to set this up in AsteriskNOW?
Thanks,
I'm number three on the dev team and not the soul person behind
FreeSWITCH. Its very uncalled for. You are dragging our project
thru the mud now also. Don't pass judgement on me. You sound quite
childish and waste my time. Never judge a man till you walk a day in
his shoes.
/b
On
Well hopefully people can read between the lines.. I have talked
about this issue in public many times and don't try to hide it but
the plea isn't how it went down.
/b
On Oct 9, 2007, at 1:50 PM, Steve Totaro wrote:
Yes, I knew who I was talking to and now I know a little more about
you
Hi, I would like to develop a click to talk app to interface with
asterisk, anyone know about some SDK/frameworks to implement this.
Regards.
Ricardo Meléndez Rosales
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On 10/8/07, Forrest Beck [EMAIL PROTECTED] wrote:
I was told that Asterisk was supported when we looked at the service.
Hey Forrest - thanks for the information. Might you be able to send
along the contact information for the TW rep who told you that
asterisk was supported? I've been in
Nick Couchman wrote:
Our office does not have a PA system, and in our current phone system we
have a certain extension that we dial that pages over the speaker of all
the phones in the office. Does Asterisk support this feature? If so,
could someone tell me the best way to set this up
PLEASE, take the old jiaxclient code and bring it back to life! It had
so much potential.
Thanks,
Steve Totaro
Ricardo Melendez wrote:
Hi, I would like to develop a “click to talk” app to interface with
asterisk, anyone know about some SDK/frameworks to implement this.
Regards.
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop some time ago, and the book was part of the
package
HtH, Hans
___
Hello,
I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42
firmware to the latest version (1.56) through tftp.
The phone loads the .inf file, then the correct firmware file (as stated
in the ST2030S.inf), then it reboots and loops doing these same things
again and again. The
Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut:
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
What about Multi-NAT where a dedicated public IP is mapped to the
private IP of the asterisk box..
ie UA - NAT - Internet - Multi-NAT -
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote:
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop some time ago, and the book was part of the
package
The
Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez:
Hi, I would like to develop a “click to talk” app to interface with
asterisk, anyone know about some SDK/frameworks to implement this.
I have not ever used such an application, but there are several
solutions commercially
On Tue, 9 Oct 2007, Brian West wrote:
I'm number three on the dev team and not the soul person behind FreeSWITCH.
Its very uncalled for. You are dragging our project thru the mud now also.
Don't pass judgement on me. You sound quite childish and waste my time.
Never judge a man till you
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut:
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
What about Multi-NAT where a dedicated public IP is mapped to the
private IP of the asterisk box..
ie UA -
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote:
Before you put any work into this... ask yourself... what exactly are
you hoping to accomplish? There is no way one system can handle a
DS3s worth of traffic... therefore, what good would this do?
I presume you can compare it with an ETSI C3
Wow. It shows that there is a lot of ignorance in the DOJ. They
should have thanked BW, not charged him. Thanks for blowing this way
off track Matt.
Tom
At 01:32 PM 10/9/2007, you wrote:
On 16:32, Tue 09 Oct 07, Steve Totaro wrote:
For a small investment of time and money, you can setup OpenVPN and have
your own network with no NAT issues whatsoever. That would be my first
choice over IAX.
Or wait till the ipv6 branch is ready for production.
NO MORE NAT ! YAY!
--
Rafael:
Thanks for your reply.
I browsed http://www.fonetglobal.com but it seems to have local numering only
in America.
We need this service but in Europe.
Do you have this service in Europe?
The thing that we need is pretty simple.
When the user calls a normal PSTN phone# from his normal
This is the first time that I am seeing this error. Can anyone help me with its
meaning ?
pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded!
Thanks.
Dovid___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On Tue, 2007-10-09 at 15:29 -0500, Erik Anderson wrote:
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote:
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop
Google for mexuar.
Zoa
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez:
Hi, I would like to develop a “click to talk” app to interface with
asterisk, anyone know about some SDK/frameworks to implement this.
I have not ever used
On Tuesday 09 October 2007 14:32:38 Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
And your point, precisely, is what?
Someone who has a criminal record can't be a technical authority? Someone
can't have a criminal record without being a scumbag? Or perhaps that you
I have configured asterisk realtime to work with two servers and a seperate
MySQL DB.
Each sip client registers which server it is connected to in the MySQL DB. This
works great as long as the clients are
1. On the same network
2. Behind a NAT and connected to the same asterisk server as the
I don't see how this is relevant to the discussion.
Zoa
Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, *Brian West* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You apparently don't realize you're talking to. Thats ok, You
keep working on
I could not tell you in asterisknow but I use this feature with Polycom
phones on all of my installs. It is very well documented in voip-info.org
Do you have any problem with the Paging when there are say 20 phones
in the page group? We have a IP601 that is used by the receptionist
and has 2
zoachien wrote:
Google for mexuar.
Zoa
Or look at one that works with MS Windows, Linux or Apple
http://www.bicomsystems.com/products/C/P/319/382/
Senad
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
On Tuesday 09 October 2007 14:20:33 Brian West wrote:
I'm number three on the dev team and not the soul person behind
FreeSWITCH. Its very uncalled for. You are dragging our project
thru the mud now also. Don't pass judgement on me. You sound quite
childish and waste my time. Never judge
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
On 18:04, Tue 09 Oct 07, Kyle Sexton wrote:
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
On Tue, 9 Oct 2007 12:06:24 -0400, Jason Aarons \(US\) wrote:
Will this work backwards? When I'm at home instead of my cell ringing
have the home phone ring? Why would anyone give up revenue from minutes?
Most won't...at least not for while. T-Mobile is the only offer
available right
Hello Gentleman Ladies ,
On Tue, 9 Oct 2007, Tilghman Lesher wrote:
On Tuesday 09 October 2007 14:20:33 Brian West wrote:
I'm number three on the dev team and not the soul person behind
FreeSWITCH. Its very uncalled for. You are dragging our project
thru the mud now also. Don't
I would recommend doing it on a 64bit platform for sure. Not sure
Asterisk has very many linger issues on 64bit... I know I run it on
64bit without too much drama.
/b
On Oct 9, 2007, at 9:32 PM, Mr. James W. Laferriere wrote:
Please , step back form the keyboard , take a deep
I have configured asterisk realtime to work with two servers and a
seperate MySQL DB.
Each sip client registers which server it is connected to in the MySQL DB.
This works great as long as the clients are
1. On the same network
2. Behind a NAT and connected to the same asterisk server as
Now the next question is why do no LGPL Dundi libs exist?
/b
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i am configured asterisk-gui the Queue Extension Configuration but
configure and register into queue.conf :
[6]
fullname = Call Center
strategy = ringall
timeout = 5
wrapuptime = 5
autofill = yes
autopause = no
maxlen = 0
joinempty = no
leavewhenempty = no
reportholdtime = yes
musicclass =
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