I hate replying to myself but this could be useful to save in mailing-list
archives.
Thanks to Digium support, I found that Alcatel PBX sends a tranfer
capability request of 3.1Khz audio, which my PRI provider does not like.
This capability is possibly generated by phones. Using:
Hi all,I've been using global and local channel channel variables extensively in my dialplan. I'd like to know what happen to those variables after the variables no longer used. Is it possible to explicitly destroy them (free the associated memory) of the variables no longer needed?
I'm using *
That's the same as I heard, but sure wish we didn't have to reboot
that new box every morning.
I heard it was actually Longhorn Embedded VoIP version with free
automatic updates if buy 500 CALS and software assurance.
-D
On 1/23/06, Cory Andrews [EMAIL PROTECTED] wrote:
Anyone have a
Hello All,
I was wondering, is anyone using verizon's fios going into a zaptel 4
port card?
If so, has anyone experienced echo issues at all?
I am under the assumption that echo issues should be minimal on a ftth
connection, but want to confirm if this is the case.
I have some customers with
Hello All,
Does anyone know if you can start an MOH queue on an individual call?
What I mean is, for example if you have a script that you want the moh
to start with certain phrases, can it be done, or are you limited to the
standard looping audio?
It's almost like starting a stream for each
Thank you Armin,
Yes, it is a fritz card :)
I till try with the overlap settings later today.
Let me know if I can be of any assistance with debug info or anything
you need.
Regards,
Nathan.
On 24/01/2006, at 3:42 PM, Armin Schindler wrote:
Let e guess, you have an AVM card?
It is a
hi,
why cant you just playback what you want to play specifically before
going to MOH, i.e.
exten = 6000,1,Answer
exten - 6000,2, Playback()
exten = 6000,3,MusicOnHold()
sorry if i'm missing something...
-yair
On 1/24/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello All,
Does
At 09:02 PM 1/23/2006, Douglas Garstang wrote:
Content-Class: urn:content-classes:message
Content-Type: text/plain;
charset=UTF-8
You aren't making calls from one phone to another, with them right next to
each other on the same desk are you?
no.
Doug.
-Original
At 10:38 PM 1/23/2006, [EMAIL PROTECTED] wrote:
I have had the same issue. It has a lot to do with the acoustics, as
well as gain. Before I messed with the config files it sounded great,
then I fussed with them and upgraded to the latest sip, and now I also
notice this on speaker.
I would go
Hi Everybody
I am building a small ippbx network for my office
I have 6 hard ip phone's and asterisk server but
now for outging and incoming calls i want to use
gsm router instead of x100p card ... or pstn
I want my calls will goout and comethrough mobile sim card (gsm router).
My mobile service
Hi,
does anybody know if there's a iax phone running on ubuntu 5.10 which
can be used with asterisk?
Seems like kiax has got many compiling and libraries problems.
TIA
Giorgio Incantalupo
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On Tue, 24 Jan 2006, Nathan Alberti wrote:
Thank you Armin,
Yes, it is a fritz card :)
I till try with the overlap settings later today.
Let me know if I can be of any assistance with debug info or anything you
need.
Thanks for the offer, I will come back to that. Maybe you can test
Hi everyone,
I am a new one for that listsactually i have final year project on VOIP IMS ...so i want to install asterisk on my pc ...IS it possble that ...we can call on small LAN network without buying any card...i will clear my point as that...suppose i have a linux machine on which i
Hi,
I'm currently looking to run Asterisk
in the office to replace an old PBX and would appreciate a little help.
We are moving offices and will have 8 digital lines. My questions
are:
As there are 8 digital lines is this
known as PRI?
Which Digium card would be the best
fit?
Would you
Hi Dirgan,
a simple google search with 'asterisk as5300' showed
some interesting information :).
I'm using a 7200 series and just had to configure my
dialpeers for voip.
Bart
Bart van Daal
Network Operations
Van Landeghemstraat 20
9100 SINT-NIKLAAS
[EMAIL PROTECTED]
www.edpnet.be
T +32
Same here.
I followed the suggestions in README.udev
compiled with make linux26 and no trouble either.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Facundo Ameal
Sent: dinsdag 24 januari 2006 2:37
To: Asterisk Users Mailing List - Non-Commercial
It would appear you are as lazy as I was when I was in university.
Answer to first question is yes
Second question is a little hopeful for an email list,
http://www.voip-info.org/wiki/view/Asterisk+introduction is your friend
Regards
Alex
-Original Message-
From: [EMAIL PROTECTED]
Google is your friend, from http://www.voip-info.org/wiki-VOIP+Phones :
http://iaxclient.sourceforge.net/iaxcomm/index.html
http://www.asteriskguru.com/tools/idefisk_beta.php
http://www.voip-info.org/wiki/view/GnoPhone
Giorgio Incantalupo wrote:
Hi,
does anybody know if there's a iax phone
Hi,
Welcome to the group anyways.
As you have asked, that's what asterisk is all about. By 'CARD' i hope u mean a provider.
Using softphones you can acall both the clients
So get going...
Have Fun.
Dan
On 24/01/06, Sohail Arham [EMAIL PROTECTED] wrote:
Hi everyone,
I am a new one for that
i don't see any other solution. you have t orecompile either the
kernel or zaptel, I recommend recompiling the kernel because then you
can continue using the new gcc version. it is not difficult, if you
want i can give you intructions so you can do it in a minute.
reagrds,
2006/1/24, Mike
Hi Douglas,
Maybe this will help you out: http://mundy.org/blog/index.php?p=92 or this:
http://www.google.nl/search?sourceid=navclientie=UTF-8rls=GGLG,GGLG:2006-0
3,GGLG:enq=asterisk+streaming+music+on+hold+moh
Cheers,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi, all
I have a problem using voipbuster (and voipstunt) for that matter.
On all calls, voice is disconnected after 30s. Asterisk still thinks that
call is in progress and I do not get any tones, just silience. Remote party
gets normal tones for disconnection.
I have paid my 10e, so it is not
Hello
Phil,
8
Digital Lines might be a fractional PRI line, or it may be that you have 4 Basic
Rate (BRI) ISDN Lines. Is there just one line coming from the "wall" or are
there 4?
Hardware wise, there are many different cards that you can
use.
The
Diva Server range of cards have
We've been asked to add H.264 and and AAC codecs pass through support to
Asterisk.
Looking at the latest 1.2 SVN branch, I see H.264 has already been
added. Does anyone have experience of using it? Any problems encountered?
Would anyone have a how-to guide (or just hints) on adding a new
Before I bought the last batch of Digium TE2xxP cards I think the story on echo
cancellation was that it only frees up your CPU cycles.
Since we were buying dual Xeons (Dell SC1420’s) for a 15 line E1 / 50 extension
system this seemed a little pointless even with all calls being recorded and
On Mon, 2006-01-23 at 23:00 -0700, Douglas Garstang wrote:
Polycom SoundPoint 601 has 4 'lines'. :)
Actually, it has 6 'lines' :)
Needing a 4 line phone is going to decrease your choices of phones.
Why do you need 4 lines?
He probably hasn't worked out the difference
On Mon, 2006-01-23 at 20:46 -0500, Jeff Herring wrote:
Issue: horrible echo (and squeals, and underwater-like sound) on speaker
phone when calling from extension to extension.
Is it a direct call from one extension to another, or a meetme, or
something similar?
echo not present when calling
On Mon, 2006-01-23 at 16:26 -0500, Bill Gibbs wrote:
I know the Polycoms work with NAT, but you have to specify the public
IP.
No you don't, at least, I never have, and it works perfectly for me
every time I have a client who regularly moves their polycom 501
from home - work and back
On Mon, 2006-01-23 at 15:34 -0500, Franklin Webb wrote:
Basically I have phone representatives that log into one of several
queues (not using chan Agent, we log in by the extension), and
frequently these agents have to make attended transfer calls to
outside numbers. This transfer basically
Hi
I looking a good IAX service for a emerging
voip provider.
Better with a test account to try.
Thanks in advance.
roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED]
For reliable and professional DNS,
On Mon, 2006-01-23 at 12:16 -0500, Steve Totaro wrote:
Is this also true for recording of calls? Will I require licensing for
each recorded call? Will the server see a big performance hit in this
setup whether or not a license is required?
In my experience (which was using asterisk 1.0.x at
On Fri, 2006-01-20 at 21:20 -0500, Michael Miller wrote:
I have over 50 Asterisk servers geographically distributed in pairs all
connected via DUNDi. Contact me off list and I will be happy to describe
my experience.
Would love to hear about peoples experiences like this.
Also, what are the
Adam Goryachev wrote:
On Mon, 2006-01-23 at 20:46 -0500, Jeff Herring wrote:
Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker
phone when calling from extension to extension.
The squeal is called feedback and comes from the sound looping back
[EMAIL PROTECTED] wrote:
Hi,
I'm currently looking to run Asterisk in the office to replace an old
PBX and would appreciate a little help. We are moving offices and will
have 8 digital lines. My questions are:
As there are 8 digital lines is this known as PRI?
In the UK, that would be
OK...Let's be clear...
1) The phones are not physically near each other.
2) It's not feedback from speaker/mic interaction.
3) Let me repeat...The squeal is not feedback, it is more of
a chirping sound at the leading edge of the spoken word.
4) I don't have 25 bad phones...they all do the
Hello and thanks for replying!
Steve,
The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
Actully ethereal,
It is encouraging to hear that
Good day.
I'm trying to configure termination with The Asterisk thru Cisco
AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network.
I think, I had recognise kind of problem: call is ringing in the
POTS phone (so I guess SIP signalling is working ok?), but there is
no voice in either
I use dbget to set callerid, but it's based on account code, and set
statically with the station, not the agent. Users can set callerid by
dialing a function coded in the dialplan for that purpose. Overhead is
not a problem.
In your case, perhaps you can set the desired callerid into a
This is entirely SIP
The behavior is only SIP to SIP...SIP to PSTN or PSTN to SIP = OK
When one or both use speaker phone, the behavior is present.
Both Handset or Headset = OK.
At 07:59 AM 1/24/2006, you wrote:
Jeff Herring wrote:
OK...Let's be clear...
1) The phones are not physically near
I think they are both great products, and we have many customers using both
successfully. You will probably be happy with either.
Both have great sound, both work well with Asterisk.
The Grandstream is easier to configure, the Sipura has more options.
More Grandsreams show up DOA, more Sipuras
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have an amportal howto for debian sarge at
http://www.squishychicken.com/index.php?option=com_contenttask=viewid=13Itemid=2
enjoy...
Dane Reugger wrote:
Sounds like good advice - I will. But would prefer to settle on Debian -
I have a how two
So: Grandstream is easy and Sipura is more flexible and complete.
Am I right?
2006/1/24, The VoIP Connection [EMAIL PROTECTED]:
I think they are both great products, and we have many customers using both
successfully. You will probably be happy with either.
Both have great sound, both work
Probably way out of line on this, but have you tried downgrading the
firmware to see if that has any impact whatsoever? (Just as a step
intended to eliminate possibilities however remote it might be.)
This is entirely SIP
The behavior is only SIP to SIP...SIP to PSTN
Sipura and Grandstream are definitely the most popular, but there are
others. There is a new IAX adapter with built-in NAT router coming soon
that might work for you. Should be announced this week. Contact me if you
think you might be interested.
Michael Crown
Managing Partner
HI,
Has anybody got any experience of the Nortel IP 2001?
We've been asked to quote for a system using ~ 250 of them, are there
any gotchas?
Cheers
Pete
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To
Facundo,
If everything goes right, we will be demonstrating an Asterisk based
Videoconferencing system at the Internet Telephony expo this week. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Facundo
This is entirely SIP
The behavior is only SIP to SIP...SIP to PSTN or PSTN to SIP
= OK When one or both use speaker phone, the behavior is present.
Both Handset or Headset = OK.
How about trying with different codecs?
___
--Bandwidth and
Hi,
I'm using asterisk 1.2.1 on a debian sarge distro.
I've followed notes in
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
and
http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO
but I still cannot change ring style via asterisk using
O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps
telling me while startup:
[chan_capi.so] = (Common ISDN API for Asterisk)
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
contr1
Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused
On Mon, 23 Jan 2006, Kevin P. Fleming wrote:
Greg Boehnlein wrote:
(Steve Totaro wrote:)
What I would really like to do is have one D channel coming in on the T3
and have it split between each of the T1/PRI or even better one D
channel per quad (I know Asterisk can do that).
Is it
I have a problem compiling bristuff modules on Fedora Core 3. Compiling
zaphfc, qozap and cwain I get thin warning:
*** Warning: zt_register [/usr/local/src/bristuff/qozap/qozap.ko]
undefined!
*** Warning: zt_receive [/usr/local/src/bristuff/qozap/qozap.ko]
undefined!
*** Warning: zt_transmit
But I'm in Argentina...
2006/1/24, The VoIP Connection [EMAIL PROTECTED]:
Facundo,
If everything goes right, we will be demonstrating an Asterisk based
Videoconferencing system at the Internet Telephony expo this week. -Mike
Michael Crown
Managing Partner
www.thevoipconnection.com
I have many Poly installations and have not had this issue, EXCEPT -
the one time we permitted a customer to run their computer data
through the telephones.
We also have noticed a poor server config can cause this in testing.
Noticed when I had one person building * servers using Debian.
Take a look at cid_rewrite from www.generationd.com This automates name
lookup (based on reverse phone number) using 411.com, a local database of
callerID's, ability to block users based on a parameter, etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Claudio Beffa schrieb:
O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps
telling me while startup:
[ISDN2]
isdnmode=ptp
isdnmode=did
might work ...
msn=51
msn= is not needed anymore. use SetCallerID in the dialplan instead.
incomingmsn=251
controller=4
softdtmf=1
Title: Asterisk with SuSe 10
Has anyone had any experience with the Asterisk on a SuSe 10 platform? I'm currently using FC3 but because we use SuSe within other parts of the business I'm being pushed to changed the OS.
Regards
Lee
###This message
The contents of the capi.conf are wrong for the version of chan_capi you use
(see capi.conf example in chan_capi package)
Or update to newer chan_capi 0.6.*
Armin
On Tue, 24 Jan 2006, Claudio Beffa wrote:
O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling
me while
I had this problem too, to fix it I had to add
[general]
vmexten=12345 ; extension to match in extensions.conf, default
'asterisk'
fromdomain=192.168.1.2 ;ip address of server, without this the voicemail
address asterisk passed to the phone was '12345@' and no domain part so
the phone just
My reccomendation. if it's not broke, then don't fix it. Unless
your getting in al new hardware, then maybe do the switch, but until
then, tell your boss, or whover is calling the shots, that the system
works, and until there is a major flaw, or a major reason to switch
other, other than just
Does anyone know if it is possible to use Microsoft Office Communicator 2005
(not Messenger) as SIP client?
I was unable even to configure!
I know it supports only TCP (and TLS) but I solved this putting SER in front
of Asterisk.
Thanks
Mimmus
___
Anybody know a tutorial , in how to create agi's? in php or other language??
thanks!
Vladimir
__
Visita http://www.tutopia.com y comienza a navegar más rápido en Internet.
Tutopia es Internet para todos.
___
--Bandwidth
On Tue, January 24, 2006 12:09, RumaTech said:
Hi, all
I have a problem using voipbuster (and voipstunt) for that matter.
On all calls, voice is disconnected after 30s. Asterisk still thinks that
call is in progress and I do not get any tones, just silience. Remote
party
gets normal tones
I found this to be very well documented:
http://asterisk-java.sourceforge.net/
I've no idea about PHP but I'm sure Google would reveal some valuable hints.
hth,
cristi
On 1/24/06, Vladimir Montealegre [EMAIL PROTECTED] wrote:
Anybody know a tutorial , in how to create agi's? in php or other
We've been trying unsuccessfully to connect our Meridian Option 81C to a
TE110P via PRI. We've followed the directions in
asterisk-meridian-a1.pdf (link on
http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it
doesn't seem to work on our 81C (even though many, many users report
Mimmus wrote:
Hi,
I know it is a FAQ but I'm interested in latest news (if any...) about SIP
over TCP support in Asterisk.
I found this:
https://savannah.nongnu.org/projects/asterisk-tcp/
but I'm not able to understand if project is active and what is its level of
development.
Thanks
Mimmus
Hi
Does anyone know a UK Voip Proivder that will give me more than 1 telephone
number and point it to my sip account.
www.SipGate.co.uk are great but they only allow 1 telephone number per user,
you can register another telephone number by registering as another user but
Asterisk doesn't
The phpagi site and package itself contains some examples that are really
helpful
scoot over to http://phpagi.sourceforge.net/ and grab it
kr,
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cristian
Draghici
Sent: dinsdag 24 januari 2006 16:04
Hi guys,I want to leave messages on our unity box. I have already converted a couple 7940s to SIP, but I can't give them out to our users because I don't want to have to deal with two voicemail systems.
we have licenses for all our users on unity as is. we're about to buy a bunch more 7940s, but I
We will Scott.
http://www.esms.com or drop me a mail off-list.
Kind regards,
SimonOn 1/24/06, scott [EMAIL PROTECTED] wrote:
HiDoes anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account.www.SipGate.co.uk
are great but they only allow 1
I setup Slimserver to stream online radio stations to asterisk. Not all
online stations work, but once you find a couple good ones you just
stick with it.
Kyle
Douglas Garstang wrote:
Has anyone managed to set up a moh server for Asterisk? Reason would be to
offload processing off the
Nobody seems to use txfax or does nobody have any problems with it?
I have sent mails to most lists and get no reply.
I cannot get a fax to go through with txfax.
I use a call file as a test and all I get on the receiving fax is a bunch of
vertical lines.
my call file is:
Hi,
On 1.0.9, an old handset with pulse disk was working and now with 1.2.2 is
not working. I used pulsedial=yes on the specific channel, but it seems to
be only for fxs signalling. How do i enable it on v1.2.2?
Regards
-=Raul=-
___
--Bandwidth and
Hi,
Maybe buy 7912 phone and register to CCM is another choice.
or integrated CCM with asterisk voicemail system.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sys
readSent: Tuesday, January 24, 2006 11:28 PMTo: Asterisk
Users Mailing List - Non-Commercial
On 1/16/06, Thczv F. Thczv [EMAIL PROTECTED] wrote:
I'm using Sipura 3000 as well, however I will have to wait until
Monday about the Switch I'm not sure. So far it looks like Sipura is
at fault. In the mean time I would like to hear from others using the
Sipura 3000 FXO if they have
Thanks Kyle. I'm wondering what the legal requirements are for streaming radio.
I guess that's one of those barrel full o' monkeys questions.
-Original Message-
From: Kyle Hagan [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 24, 2006 8:36 AM
To: Asterisk Users Mailing List -
On Tuesday 24 January 2006 09:26, Lee Archer wrote:
Has anyone had any experience with the Asterisk on a SuSe 10 platform?
I'm currently using FC3 but because we use SuSe within other parts of
the business I'm being pushed to changed the OS.
Just about all of my production Asterisk servers are
I have my eyes on the Linksys/Sipura 941, ( SIP ), but the core problem is that you can't use SIP phones with CCM. I have a SIP trunk between asterisk and ccm. I can route calls back and forth, I just can't get the call to send to vm if no answer on the asterisk side.
On 1/24/06, kevin ling [EMAIL
I have an Asterisk 1.2 install running on RedHat 9. I have a bunch of
Polycom 501s co-locacted in the same building as *, and some more
501s in satellite offices (also registered to my * server) . Finally
I have some road warriors running XLites.
Ideally when a road warrior (XLite) calls a
Thanks, I've got it running on my test box but didn't know if there was
any global objection to using it. I've had a few funnies with it but
that might be down to Supermicro and P4's with the EM64T thing.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
You left the attribution off the quote you included with your mail.
That was Dovid Bender, right?
Brian, Yes, sorry... I believe so, yes.
Ira, you're right. At the time it was available it was a monumental
guide. Maybe there will be a newer version RSN? Still, for a number of
questions about
Downgrade your spandsp. Do some reading on spandsp first!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Allan Gee
Sent: Tuesday, January 24, 2006 10:36 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] txfax application problem
Nobody
Hi
all,
i need to install
chan_h323, just a question: to use it i need the CVS ?
Is therea
version that i could use with asterisk 1.0.9 ?
Thanks
Giordano
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China: The World's New Engine for Growth
Everywhere smart investors have been accumulating positions in China
related equities, with the knowledge that this vast new market is no doubt
the place to be for eye-popping profits.
And today's profiled company Ever-Glory International Group, Inc
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On Tue, Jan 24, 2006 at 09:13:45AM +, scott wrote:
Does anyone know a UK Voip Proivder that will give me more than 1 telephone
number and point it to my sip account.
www.SipGate.co.uk are great but they only allow 1 telephone number per user,
you can register another telephone number
You can try Voxboned(www.voxbone.com) if you need inbound only.
On Tue, 2006-01-24 at 09:13 +, scott wrote:
Hi
Does anyone know a UK Voip Proivder that will give me more than 1 telephone
number and point it to my sip account.
www.SipGate.co.uk are great but they only allow 1
In sip.cfg, add something like this:
alertInfo voIpProt.SIP.alertInfo.1.value=ring3
voIpProt.SIP.alertInfo.1.class=4/
...to correspond to something like this...
SPECIAL_RING se.rt.4.name=ring3 se.rt.4.type=ring-answer se.rt.4.timeout=2000
se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/
On Tue, 24 Jan 2006, scott wrote:
www.SipGate.co.uk are great but they only allow 1 telephone number per
user, you can register another telephone number by registering as
another user but Asterisk doesn't allow multiple registrations.
Don't be silly, of course it does - I have about 4
I was wondering if there is anything new in 1.2.2 for zap channels (analog)
that can detect when a person answers the call?
When I place calls in a call file (just to speak the demo-congrats
message) for an
analog channel it dials and starts playing the mesage before I even
answer the phone.
I was wondering if there is anything new in 1.2.2 for zap channels (analog)
that can detect when a person answers the call?
When I place calls in a call file (just to speak the demo-congrats
message) for an
analog channel it dials and starts playing the mesage before I even
answer the phone.
I was wondering if there is anything new in 1.2.2 for zap channels (analog)
that can detect when a person answers the call?
When I place calls in a call file (just to speak the demo-congrats
message) for an
analog channel it dials and starts playing the mesage before I even
answer the phone.
Hello
Can we find a patch for asterisk-1.2.2 in order to
test ACD with polycom phones ?
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs
I was waiting for the spam to start... Looks like
were off
--- Laurie Whitaker [EMAIL PROTECTED]
wrote:
China: The World's New Engine for Growth
Everywhere smart investors have been accumulating
positions in China
related equities, with the knowledge that this vast
new market is
We have just analog lines coming in to our Asterisk box and so no
CallerID information can be gathered, all calls look the same on the
phone display.
Once a user parks a call and the time runs out it returns the call but
keeps the original CallerID information that makes it look like it is
just
You can have asterisk dial your Unity vmail pilot on busy or
unavailable, and have CCM use the last redirected number on the trunk to
determine the called extension, or pass the $RDNIS value and digit
add/strip from * to CCM.
We use * in the exact opposite fashion, but should suffice either
I have lots of accounts registered but cannot get asterisk to register and
recieve calls for those accounts.
It appears to register one account but I can only ever get incomign calls from
one number.
voip-info.org also states that asterisk cant handle multiple registrations??
thanks
scott
If you have 16 call appearances or lines - how do you get to line 16 -
type in some code?
Adam Goryachev wrote:
On Mon, 2006-01-23 at 23:00 -0700, Douglas Garstang wrote:
Polycom SoundPoint 601 has 4 'lines'. :)
Actually, it has 6 'lines' :)
Needing a 4 line phone is
Why do you need a patch? We have ACD/Asterisk 1.2.1 working well with Polycom
IP phones. Haven't done much with 1.2.2 yet. Is there some sort of issue?
About the only thing that doesn't work is the appearances don't display the
login/logout status with the icon of an agent in an ACD Queue.
right, but how do I pass the rdnis to ccm?On 1/24/06, Greg Oliver [EMAIL PROTECTED] wrote:
You can have asterisk dial your Unity vmail pilot on busy orunavailable, and have CCM use the last re
directed number on the trunk todetermine the called extension, or pass the $RDNIS value and digit
I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data
package. I continue to pay for a standard voice line in addition to the
broadband connection only for local calling, fax and emergency 911 use.
The way it works is that the fiber optic connection is terminated on the
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