Otherwise,
The official asterisk GUI will NEVER work in 1.2, since the manager over http
has not been, and never will be in (in the real branch) because 1.2 is no longer
being committed to, and even if it was, its not a bug fix, so it would not
anyways.
-bk
- Original Message -
From:
Hi ,
I am trying to dial in from two sip phones on one end, through
digium card to E1 card running application on another end.
with following configuration
/etc/asterisk/zapata.conf
group=1
context=default
euroisdn=EuroISDN
signalling= pri_net
Hi guys,
I spent a couple of hours in Google, but the problem
appears to be uncommon, so I'd like to ask about it here.
The problem is exactly the opposite to Asterisk does
not detect FXO hangup. In my case it's the Telco who
does not appear to be detecting Asterisk's hangups.
Telco is Telus
Hi:
Please every that work with A102d say how about is it?Is it really difficult to
install card for me new in asterisk?
Best regards.
-
Boardwalk for $500? In 2007? Ha!
Play Monopoly Here and Now (it's updated for today's economy) at Yahoo!
Hello,
To connect Asterisk to Frame relay network, have i to use the wildcard
TE110P.
Thanks.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Mon, 6 Aug 2007, MOSBAH ABDELKADER wrote:
To connect Asterisk to Frame relay network, have i to use the wildcard
TE110P.
As long as it supports frame relay encapsulation (it appears to), sure.
But what do you mean by connect? Even if you must use frame relay,
why insist on TDM?
Hi to all,
I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.
I've tested h323 using ohphone and I can talk between them, then I've tested
SIP with Twinkle softphones and function very well.
Now I have to perform call from h323 to sip and viceversa.
How can I do it
I
Hi All,
The Asterisk book (the pdf version) is excellent!! I want to thank all the guys
that put it together. I am most grateful for it.
There is something about writing a dialplan that I'm not clear about. What I'm
trying to figure out how to do is this: when I transfer a call to the
That's ok, and is expected behaviour. The telco will keep the line
open for about 30 seconds. It's useful when there is no PBX, and just
2 or 3 phones attached to the same line... you can hangup on one room,
go to another, pickup and continue the conversation.
Anyway, i guess the telco can reduce
Hi Alex,
You should create a dial plan to route sip calls to H.323 calls.
Take a look at :
http://www.voip-info.org/wiki/
On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi to all,
I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.
I've tested h323 using ohphone
Hi,
thanks for reply
I'm reading more about Dialplan, but until now, I've not found
anything...(like example or tutorial)
With the word route you are intending the Goto command??
Please spent some minutes for helping me ^_^
If you are agree, I send you some information about configuration files.
Ok thanks looks like the card is running the most recent version
of the firmware.Oh well.. as you said Sangoma tech support is
wonderful... and I have no doubts they will assist in resolving the
issue.. just wish they weren't PST (or some variation of that).
On 8/5/07, Tom [EMAIL
Hi, Sangoma is in my opinion the best card for asterisk, better until
Digium, is practically plug in play and also the support technician found
in wiki in the site of the Sangoma is very good. It sees:
http://wiki.sangoma.com/wanpipe-linux-asterisk
Regards
Josué
2007/8/6, fateme fatah [EMAIL
Hi all,
I got serious problem here, I hope I ask on the right place here (sorry if I
am wrong).
I have used asterisk 1.2.17 with openh323 ver. 0.7.3, for integrating
between SIP Gateway and H323 Gateway, it runs about 6 months. But, recently
I think it doesn't work anymore...I can't call from
On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote:
In the design of an Asterisk system using Cisco 7900 series SIP phones
we are struggling with giving the reception folks (3) hardware that can
tell them the status of everyone in the office (10 or so) (On the phone,
out of office etc)
Hi Alex,
you should have a route for each extensions you would like to reach in
your extension.conf file.
Dial Plan is the main concept to understand in Asterisk.
Feel free to send you conf and I'll take a look.
On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi,
thanks for reply
I'm
On 8/3/07, Michael Munger [EMAIL PROTECTED] wrote:
Is there a way to setup an IAX bat phone (immediate=yes) or is this a
privilege only reserved for ZAP channels?
As I understand it, this would have to be supported by your specific
hard/soft phone.
It's the same with SIP - taking a handset
With Polycom 501s, creating custom ringtones isn't hard at all.
First, grab your favorite mp3 or wav file and create a file that is
about 10 seconds long (max). If its an mp3, convert it to a wav file.
Next, use this command to ensure the wav file is properly formatted for
a Polycom phone:
sox
This is an option in the queue application. You could just create a
queue for that single user.
Thanks,
Steve
Jeng Yu wrote:
Hi All,
The Asterisk book (the pdf version) is excellent!! I want to thank all
the guys that put it together. I am most grateful for it.
There is something about
If you setup your dialplan correctly and use priority jumping (off by
default), you can play congestion or you could play the unavailable
message for voicemail an then have the person who's phone it is record a
message saying that they are on the phone, please leave a message.
Thanks,
Steve
Michiel van Baak wrote:
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
WaitExten is waiting for
Stephen Bosch wrote:
Steve Totaro wrote:
Note to Digium
I wish I could upgrade my wct4xxp drivers locally. I still have the v1
firmware on my card.
It is kind of hard (next to impossible) to pull it from a production
machine and ship it to Digium. That might take a week if all goes
Hi,
i am having problem on playing asterisk sound file on my new installed
asterisk..
i have the following extension , if i call from any SIP / IAX phone
playback or voicemail doesnt play anything but when i dial 102, I
hear the MP3 music ..
exten = 99,1,Answer()
exten =
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
Getting even close is hideously expensive.
On Mon, 2007-08-06 at 10:05 +0100, Jeng Yu wrote:
The Asterisk book (the pdf version) is excellent!! I want to thank all
the guys that put it together. I am most grateful for it.
I'm glad you enjoyed the book.
I want to play a greeting message to the person called for a few
seconds before
Hello everybody,
I'm aware that I can try to balance gain levels with PSTN cards using the
ztmonitor tool, as described in
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html
(Adjusting the rxgain/txgain Settings).
Is there a similar tool for mISDN? If
On Sat, 2007-08-04 at 12:41 +0300, Mustafa Sakalsiz wrote:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 1ms SCall: 4 DCall: 3 [10.10.10.73:4569]
CAUSE : No authority found
CAUSE CODE : 50
This no authority found
On Mon, 2007-08-06 at 09:50 -0400, Steve Totaro wrote:
If you setup your dialplan correctly and use priority jumping (off by
default), you can play congestion or you could play the unavailable
message for voicemail an then have the person who's phone it is record a
message saying that they
Hi,
I have an 8-port Grandstream GXW-4008 V1.2A ATA
converter with analog phones connected to it.
They work fine except for just one feature I would
like to modify. Somehow, each time the ATA
re-registers the SIP clients or each time the device
has to be rebooted for maintenance, the phones ring
i want more example of extention.conf i have find many on google but it is
documented i want live example if u have extention.conf can u send me working
extention.conf i m new for asterisk so that send me one file
Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-08-06 at 09:50 -0400, Steve
i am new for asterisk can u give me suggestion for my setup i m not expert for
dialplan so can u send me example file which one u have i need help for
extention.conf file options
Rgds
satish patel
Steve Totaro [EMAIL PROTECTED] wrote: If you setup your dialplan correctly
and use priority
MOSBAH ABDELKADER wrote:
Hello,
Have i to buy an asterisk card like TDM400P to connect the two asterisk
servers with frame relay.
I never do that. I use a router that supports Frame Relay. For me,
installing a Digium card just to connect to a Frame Relay network is
much more work, poorly
Douglas Garstang wrote:
I don't think creating a network without a single point of failure is
unreasonable.
How often have you built a network without a single point of failure?
___
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Douglas Garstang wrote:
Let's assume for a moment that it's impossible. That does not mean adding
additional servers and additional networking equipment does not add value, or
is a worthless endeavour.
I agree with that. At least two people that I know run ITSPs. Each
time they have an
Flash Operator Panel would do it.
Also the Aastra 55i phones with the expansion module, which has 36 lines on
it should work, but you will need to cofigure your Asterisk for Shared Line
Appearances (also called Bridged Line Appearance) for the Busy Lamp Field
(BLF) to work. The Aastra 55i would
Hi,
Does Asterisk support rfc3680 ?
This relates to registration event package.
This features seem to be convenient when implementing free sitting features
Regards
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Hi. I have a server which is trying to register with me using iax2
and asterisk 1.2. My asterisk server is rejecting the registration
saying ip address of server is not dynamic. What does this mean and
what do I need to change to accept the registration?
Thanks.
--
Your life is like a penny.
On Mon, Aug 06, 2007 at 12:18:36AM -0400, SIP wrote:
Jay R. Ashworth wrote:
ASCAP and BMI annual blankets aren't actually that expensive. A live
music venue run by some friends of mine had both, and for 535 fire-code
seating and about 150 nights a year, I think they paid $500 a year to
Hi,
you must choose the h323 channel, install and configure it.
I've used ooh323 for a similar project and I have edited the
ooh323.conffile with the Gnugk's IP address (your h323 gatekeeper) and
a new context
for your test. I've also configured the file .ini in Gnugk (I've used the
Win version).
Jay R. Ashworth wrote:
However, if you get caught willfully performing copyrighted music
without paying ASCAP, BMI, et al, you're liable for a $100,000 fine
($20,000 per song if it's not deemed willful) per song.
I wonder how much of *that* money goes to the songwriters. ;-)
Cheers
On Mon, 2007-08-06 at 11:00 -0400, John covici wrote:
Hi. I have a server which is trying to register with me using iax2
and asterisk 1.2. My asterisk server is rejecting the registration
saying ip address of server is not dynamic. What does this mean and
what do I need to change to accept
On Mon, 2007-08-06 at 07:42 -0700, Vieri wrote:
What I don't know yet is if it's a purely ATA
config-related issue or if I also need to change
Asterisk's settings.
As far as I know, this is a setting on the ATA, and nothing you change
in Asterisk would affect it.
--
Jared Smith
Community
Hello,
How would you implement free sitting ?
The idea is to offer teachers the ability to share the same desk and
hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr
Bar is teaching english on wednesdays.
Each has his own extension but use the same hardphone.
1. Does a
Thank you for your reply as it is exactly what we would need. Sorry I
didn't find it myself. I do have a question about configuration within
Asterisk.
I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated ...'
Is that what we are
OK, thanks -- I guess I hadn't quite figured out the purpose of
registration.
on Monday 08/06/2007 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2007-08-06 at 11:00 -0400, John covici wrote:
Hi. I have a server which is trying to register with me using iax2
and asterisk 1.2. My asterisk
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a single example where this
actually exists.
On 8/6/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
I have an 8-port Grandstream GXW-4008 V1.2A ATA
converter with analog phones connected to it.
Hi,
we hava a GXW-4004 but i think it has the same sw ...
They work fine except for just one feature I would
like to modify. Somehow, each time the
Freepbx has devices and users concept. It may be what you're looking for.
You can have your users log in in any phone with their extension
number and password. After that, all calls to his extension would ring
on that phone.
http://www.freepbx.org
Julian J. M.
On 8/6/07, Olivier [EMAIL
On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
OK, thanks -- I guess I hadn't quite figured out the purpose of
registration.
A device registers to Asterisk to tell Asterisk what it's current IP
address is, so that Asterisk knows where to send calls destined for that
device. That's all
Steve Totaro wrote:
Anthony Francis wrote:
Tim Panton wrote:
On 5 Aug 2007, at 06:54, Douglas Garstang wrote:
I don't think creating a network without a single point of failure
is unreasonable.
It's impossible. I can't think of a
On 8/6/07, James R. Stevens [EMAIL PROTECTED] wrote:
I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated …'
Is that what we are doing within Asterisk or Trixbox when we configure an
extension? (A Directory Number??)
I suspect DN
Hi,
I'm trying to use a Detewe TA 33-clip but there is no rj11 connector on
it...only a TAE connector.
I'd like to create an adapter so I need to know which TAE pins to
connect to RJ 11 pins.
Is there anybody who knows where I can find a schema of that adapter?
Single connector pinout may help
Stephen Bosch wrote:
The only way this will ever happen is if Digium completely redesigns the
card, which is a long way of saying that you will buy a new card before
you have that request filled.
That is incorrect. The TE4XXP cards with v2 or later firmware *can* be
upgraded in the field, but
Kevin P. Fleming wrote:
Stephen Bosch wrote:
The only way this will ever happen is if Digium completely redesigns the
card, which is a long way of saying that you will buy a new card before
you have that request filled.
That is incorrect. The TE4XXP cards with v2 or later firmware
The 7914 only works under SCCP; the SIP firmware does not support it at
all (the expansion panel won't even power on fully.) The SCCP channel
driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been
much discussion
And what of all the folk that have a v1 card (I've got 2 quad-ports
sitting here) ?
And can you cross-ship a v1 card for a v2 card replacement ?
Julian.
Steve Totaro wrote:
Kevin P. Fleming wrote:
Stephen Bosch wrote:
The only way this will ever happen is if Digium completely
Eric ManxPower Wieling wrote:
Douglas Garstang wrote:
Let's assume for a moment that it's impossible. That does not mean adding
additional servers and additional networking equipment does not add value,
or is a worthless endeavour.
I agree with that. At least two people that I know run
Might as well unless you have to pay shipping twice.
Wanna sell one of those quad port cards if it is just sitting there
(after you get the firmware upgraded of course :-) )?
Thanks,
Steve
Julian Lyndon-Smith wrote:
And what of all the folk that have a v1 card (I've got 2 quad-ports
sitting
Kevin P. Fleming wrote:
Have you looked at the Sangoma cards and the Digium cards? Did you
notice that *both* of them are based on large Xilinx FPGA parts? They
both use an 'FPGA architecture', at least for the PCI interface and
TDM/data buffering (both cards use dedicated T1/E1/J1 framer
On Mon, 2007-08-06 at 17:46 +0200, Mr Shunz wrote:
we had the same problem and we came to this solution:
go under profile settings and set
Caller ID Scheme as
ETSI-FSK Prior to Ringing with DTAS...
best regards
I'm experiencing the same issue with linksys pap2. Any knows how to stop
(top-posting because Julian did, and I'm too lazy to fix it all)
Last I checked, the replacement with the new firmware is only for those who
bought the card in the last year (i.e. the card is still under warranty).
Those of us who were early adopters cannot enjoy the improvements of the
upgraded
Too bad it is August 6th
*P.S. Remember, as a member of the Digium Family we have secured a
special discount of 50% off of the conference fee for you if you
register by July 29, 2007. To take advantage of this limited time offer,
please register here
On Mon, Aug 06, 2007 at 11:26:25AM -0400, SIP wrote:
I actually tried to find that out (even something anecdotal), but so far
no luck. I'm guessing not that much. The law allows for adjusting the
percentages somewhat on the fly for various reasons (for instance, web
radio performances give
On 6 Aug 2007, at 16:53, Jared Smith wrote:
On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
OK, thanks -- I guess I hadn't quite figured out the purpose of
registration.
A device registers to Asterisk to tell Asterisk what it's current IP
address is, so that Asterisk knows where to
Tim Panton wrote:
On 6 Aug 2007, at 16:53, Jared Smith wrote:
On Mon, 2007-08-06 at 11:42 -0400, John covici wrote:
OK, thanks -- I guess I hadn't quite figured out the purpose of
registration.
A device registers to Asterisk to tell Asterisk what it's current IP
address
All,(Ryan-Your response saved me lots of RD time- Thank you very much)
I have been on TixBox site all morning reading through the MANY posts as
recent as 6-4-2007 on SLA and the need or reason it is not needed.
1) In our office we do not have a single receptionist, rather a
ring group
Yes, since IAX2 only uses one port, this is correct. Another thing to
keep in mind is to set a low qualify value in Asterisk since some
routers will tear down the connection pretty quickly. The qualify acts
as a keep-alive and prevents the router from closing the port and losing
the map.
I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up. The bchannels
are all up and the T1 is not in alarm status. The dchannel refuses to
come up however. We've tried ni2, qsig, and now dms100 for the
switchtype. The telco
Thanks.
In fact, my questions are more about usage than about technical background.
For instance, I doubt a user will log his system off when leaving : some
don't even turn their PC off.
Does anyone has an experience to share about that ?
___
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Thanks,
Steve
Erik Anderson wrote:
I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up.
At 08:16 8/6/2007, Rob Schall wrote:
With Polycom 501s, creating custom ringtones isn't hard at all.
First, grab your favorite mp3 or wav file and create a file that is
about 10 seconds long (max). If its an mp3, convert it to a wav file.
Next, use this command to ensure the wav file is properly
This Friday, part II of TDM solutions including ATA that do IAX and
SIP without opening the box and installing a card. Your experience in
this area would be appreciated.
You can find us here:
http://www.AsteriskUsersConference.org
Also, a Google group has been created for discussions and
wanpipemon is the way to do it as far as I know.
For starters, what do your zaptel/zapata configs look like?
I would first verify that your D-channel is set properly, you can view
that in the console as follows:
asterisk pri show span 1/0
Primary D-channel: 24
Status: Provisioned, Up, Active
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal contact there (Jignesh) is either
Erik Anderson wrote:
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal
On 8/6/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
wanpipemon is the way to do it as far as I know.
For starters, what do your zaptel/zapata configs look like?
lpdlnx04*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: Nortel DMS100
Type: Network
I know
Erik Anderson [EMAIL PROTECTED] wrote:
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal contact there (Jignesh) is either out of the office today or at
least he forgot to start up MSN this morning, as he's showing offline.
Hopefully he's not the only tech support
Grandstream HT386 also has that feature. Into the configuration you
can find a field called 'Audial Off-hook', there you can set any
extension so the ATA will dial as soon as you pick up the handset.
On 8/6/07, James FitzGibbon [EMAIL PROTECTED] wrote:
On 8/3/07, Michael Munger [EMAIL PROTECTED]
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
I have done a conference call with the telco guy, myself, and a Sangoma
tech at the same time. I was just quite and let them battle it out. It
turned out to be a telco issue but the Global Crossing tech wanted to
blame me and my equipment.
Darren Nickerson wrote:
Erik Anderson [EMAIL PROTECTED] wrote:
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal contact there (Jignesh) is either out of the office today or at
least he forgot to start up MSN this morning, as he's showing offline.
Hopefully
Erik Anderson wrote:
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal
Olivier wrote:
Thanks.
In fact, my questions are more about usage than about technical
background.
For instance, I doubt a user will log his system off when leaving :
some don't even turn their PC off.
Does anyone has an experience to share about that ?
Hi David -
Last I checked, the replacement with the new firmware is only for those who
bought the card in the last year (i.e. the card is still under warranty).
Those of us who were early adopters cannot enjoy the improvements of the
upgraded firmware without buying all new cards.
Thanks, Julian.
I saw this explanation, and it does not apply. There is no
hangup supervision with my carrier. I think they used to
have it, when I had different number, but even then it was
not 30 seconds, but more like 3 to 5.
I am now inclined to think that it has something to do with
a
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to
them.
I see plenty of SABMEs, but nothing else:
[ 02 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000
Hi,
I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The
add-ons pack has been installed for a while, so now I'm trying to add
the Mysql config.
I've created a mysql database, added the grants for a user acces, and
can run a mysql -u asteriskcdruser -p and can connect to the
David Gomillion wrote:
Last I checked, the replacement with the new firmware is only for those
who bought the card in the last year (i.e. the card is still under
warranty). Those of us who were early adopters cannot enjoy the
improvements of the upgraded firmware without buying all new cards.
Kevin P. Fleming wrote:
David Gomillion wrote:
Last I checked, the replacement with the new firmware is only for those
who bought the card in the last year (i.e. the card is still under
warranty). Those of us who were early adopters cannot enjoy the
improvements of the upgraded firmware
Have you completely ignored the telco suggestion and attempted pri_cpe?
Sounds like a miscommunication in settings to me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Anderson
Sent: Monday, August 06, 2007 12:02
To: Asterisk Users Mailing List -
Darryl Dunkin wrote:
wanpipemon is the way to do it as far as I know.
For starters, what do your zaptel/zapata configs look like?
I would first verify that your D-channel is set properly, you can view
that in the console as follows:
asterisk pri show span 1/0
Primary D-channel: 24
Status:
Erik Anderson wrote:
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to
them.
I see plenty of SABMEs, but nothing else:
[ 02 01 7f ]
Unnumbered
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
You should never be the signaling source, you are always a slave to the
provider, go with pri_cpe and see if things go better.
That's what I've experienced in the past, but they were adamant about
me being the network end. I tried switching
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Monday, August 06, 2007 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Bosch
Sent: Monday, August 06, 2007 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Eric ManxPower
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
Yeah you are sending the SABME's because you think you are the master,
they are not replaying with a UA because they think they are the master,
you should def be pri_cpe.
Tried it...no go.
There is one other potential cause here, you may
Hello
I need a Softphone with auto answer where users can't turn it off.
Does someone knows a softphone where users can't turn the auto answer off?
Or is there any way Asterisk could force the clients to answer the phone?
Thanks
Regards
Joao Pereira
I'd like to know what alternative is available for those who run a
call centre with dynamic agent-queue allocation.
We have people monitoring the queues and assigning agents depending on
the queue demand.
cheers!
Santiago
On 7/5/07, Martin Schrott - thinking:systems [EMAIL PROTECTED] wrote:
In fact, my questions are more about usage than about technical background.
For instance, I doubt a user will log his system off when leaving : some
don't even turn their PC off.
Does anyone has an experience to share about that ?
When I tried it, when a user login at a phone, it replaced
Time Bandit wrote:
In fact, my questions are more about usage than about technical background.
For instance, I doubt a user will log his system off when leaving : some
don't even turn their PC off.
Does anyone has an experience to share about that ?
When I tried it, when a user login
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of SIP
Sent: Monday, August 06, 2007 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
Steve Totaro wrote:
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