Doh, I didn't compile meetme support into this install because it
said it required a zaptel card for the hardware timers. I have a
spare zaptel card, but I'd rather not install it if I don't have to.
I didn't know of the ztdummy module, I will compile this and then
recompile asterisk.
Hi
I've few queues configured in * box is there any what that before sending
call to a particular queue can we get the status of the queue that is how
many agents are available in this queue (logged in, paused, busy,
unavailable).
thanks
arun
___
Here is some more details about my setup:
Customer - PRI - Server A with G.729 - IAX Trunk G729 - Server B no
G729 (pass through) - Snom Phone with G729
with incoming call there is no problem with when I try to make outbound and
want to play some prompt on server b Im not able. in server B
I like to forward them back to themselves, that is, the ones that give
their phone number. Check nerdvittles.com. I think he had some kind
of torture script setup, if I remember correctly.
On 5/5/07, Salvatore Giudice [EMAIL PROTECTED] wrote:
Just forward them to 1-800-big-dick or some other
HI
I have 3 Linksys SIP901 IP phones
I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I'm looking to connect this phones together and to make calls between them
Not from outside of my lan
I don't know how to configure asterisknow beta
Can somebody help
I'm
Hi Guys,
I have some Polycom 601's here. It is super annoying that the phone every so
often beeps to let me know that I have a VM. Is there any way to turn that off
? (I just want the red led to blink that there is a VM).
Thanks.
Dovid___
--Bandwidth
Hello,
When dialling a SIP phone which is already in a call the caller hears a
regular ringing tone and does not know that the called party is engaged in
another call. Is there a supported way inside SIP to tell the calling party to
play a stuttered ringing tone?
I've found some manuals and tried this to do :
Sip.conf
[test]
type=friend
username=test1
secret=test1
host=192.168.1.238
context=tutorial
fromuser=SIP Phone
callerid=101
nat=no
canreinvite=yes
dtfmode=info
disallow=all
allow=ulaw
[test]
type=friend
username=test
First of all, both users have the same name:
[test]
type=friend
username=test1
...
[test]
type=friend
username=test
...
You should change this to:
[test1]
type=friend
username=test1
...
[test]
type=friend
username=test
...
Nico
Ardit Saliu schrieb:
I’ve found some manuals and
With Polycom phones, you should steer clear of headsets with in-line
amplifiers. We have found these to introduce electrical hum into the
audio streams.
Just an FYI.
Thanks,
Bryan Johns
Partner
Shelton | Johns
1805 Old Alabama Road
Suite 200
Roswell, GA 30076
USA
Office: 678.248.2637
Google: polycom mwi beep
-- http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
The solution given works for me...
Alvin
Dovid B wrote:
Hi Guys,
I have some Polycom 601's here. It is super annoying that the phone
every so often beeps to let me know that I have a VM. Is there
We are running 2.1.1 with 3 expansion modules. Presence in turned on in
sip.cfg, hints are set up on the extensions and buddy watch is enabled on
about 40 extensions mapped to the expansion modules. We have 3 paging
groups: Building A with 20 phones, Building B with 20 phones and Both
with all
Hello,
this is a SIP phone configuration issue.
You should tell the UAC to not accept a second call while the line is
engaged (look for a 'Call Waiting' option in the configuration of the UAC)
The UAC will send back a 486 Busy Here error code and the calling
party will get a busy signal
What logic are you using to determine if the caller is indeed a
telemarketer, anyway?
Lacy Moore - Aspendora wrote:
I like to forward them back to themselves, that is, the ones that give
their phone number. Check nerdvittles.com. I think he had some kind
of torture script setup, if I
Hi Salvatore, thanks in advanced.
I really go to try to make some tests with asterisk, I find that it will be
able to take care of not for complete, but part of this necessity. I sending
a message for the list on the results gotten in the tests.
Best Regards
Josué
2007/5/6, Salvatore Giudice
You do not need a zaptel card for ztdummy. And on the phone itself, the
extensions were still configured as private. SLA on a Polycom through
* is just presence on a line key (aka speed dial) + autoOffHook.
-Original Message-
From: Jay Austad [mailto:[EMAIL PROTECTED]
Sent: Sunday,
aaa
I have 3 Linksys SIP901 IP phones
I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I'm looking to connect this phones together and to make calls between them
Not from outside of my lan
I don't know how to configure asterisknow beta
Can
Trixbox is intended to your kind of public. You can use it, it will work for
you without any problem.
Regards,
Panlo
I have 3 Linksys SIP901 IP phones
I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd
I'm looking to connect this phones together and to make calls
this is a SIP phone configuration issue.
You should tell the UAC to not accept a second call while the line is
engaged (look for a 'Call Waiting' option in the configuration of the UAC)
The UAC will send back a 486 Busy Here error code and the calling
party will get a busy signal from
bkruse wrote:
Good question shawn,
The callfile does get deleted once the call has been finished (I believe
its FINISHED, not processed)
No, they are not executed sequentially..exactly. Well, from your
point of view, you can drop tons of them in there and all of the calls
will fire
Manually,
I'm just designating some people as telemarketers (using some MySQL +
AGI to evaluate and drop the call) I would like to have a little fun.
- Adam
On May 6, 2007, at 12:06 PM, Steve Finkelstein wrote:
What logic are you using to determine if the caller is indeed a
telemarketer,
There is a wiki article on doing it. It shouldn't be a problem at all,
in fact the wiki article is for Avaya ACS Partner. Since the Avaya
uses DTMF for VM it's quite easy to do.
On 5/6/07, Josué Conti [EMAIL PROTECTED] wrote:
Hi Salvatore, thanks in advanced.
I really go to try to make some
Hi all,
I have a hangup problem when i get incoming calls on my ISDN interface.
I use ISDN network controller [HFC-PCI] and asterisk with florz patch.
Logs when the hangup happens follows:
May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' not posted
May 6 20:52:47 NOTICE[11532]
How about a mysql sync - shared module that we could collectively poll
periodically...to track exactly thisshare our telemarker numbers
Would want 1. number, 2. label of who it is... 3. last hit on it...
We'd each track our own...maybe an enhancement to the blacklist
module...
Use a peer
As good of an idea as that sounds, it would certainly need to be
moderated in one form or another. It's just begging for abuse to allow
any user to arbitrary insert numbers on the PSTN deeming them as
blacklists. Sort of a spamhaus for telephony services ..
Dave Bour wrote:
How about a mysql
So how well does SLA work?
The FreePBX developers (well, at least one) seem to think it's a non-starter
and dumbs down the other features too much. I haven't experimented with it
yet so I'm eager to hear some real world feedback.
-Original Message-
From: David W. Rice [mailto:[EMAIL
No problem. Sorry about the delay. Unfortunately, I no longer work for a
Polycom reseller, so I can't give you a simple link to the files that
you need (someone else on the list might be able to help you out
off-list with this). Sometimes, you can find them online by searching
google with Polycom
Also,
What *is* a telemarketer?
I have the ACLU on my blacklist, i'm a big contributor to the ACLU
and instead of being happy, they take this as a sign that they should
bleed me dry further (the same for lots of political causes).
What occurs to me here is something on the order of dnsbls
Hi,
I have problem with setting up a conferences. When I dial the reserved
conference number from xlite the line gets hunged up
and on a console there is a following message:
WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe'
for extension (internal, 1234, 3)
exten =
True enough...how about this then...
For any unknown numbers since I already do call lookups on existing
numbers in my local database (Asteridex) followed by google and couple
others...anything coming back unknown...would like to find at least what
it is and call display it
So a few things
Can someone recommend a good quality 24 or greater port channel bank?
Steve
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Am Sonntag, den 06.05.2007, 00:48 -0400 schrieb Salvatore Giudice:
Just forward them to 1-800-big-dick or some other 800 toll free phone sex
line. They can't tell they've been forwarded. They'll figure it out
eventually.
Whoa, that was _my_ coffee that's now on the screen.
I will urgently
Forum wrote:
Can someone recommend a good quality 24 or greater port channel bank?
The Adit 600 is a favorite of mine.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
On 5/6/07, Doug Lytle [EMAIL PROTECTED] wrote:
The Adit 600 is a favorite of mine.
Doug
Would second the Adit too. We are running a Rhino now and have had no
problems with it.
-Brian
___
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On May 6, 2007, at 6:47 PM, Anselm Martin Hoffmeister wrote:
Am Sonntag, den 06.05.2007, 00:48 -0400 schrieb Salvatore Giudice:
Just forward them to 1-800-big-dick or some other 800 toll free
phone sex
line. They can't tell they've been forwarded. They'll figure it out
eventually.
Whoa,
On Fri, May 04, 2007 at 09:59:36PM -1000, Mark Coccimiglio wrote:
Tzafrir;
Actually I have found this config to work really well. I prefer to
use a script run from inittab but Ubuntu doesn't work like Redhat or
BSD. On a production box keeping asterisk up and running is THE TOP
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