Hi there,
is there any way to call Asterisk from a SIP phone, where you don't know
name and password of the caller?
I want to allow customers of a company to place a call over the Internet
without being registered on the Asterisk. This could be a very large number
of SIP clients, only a few will
Hello richard,
Wednesday, November 23, 2005, 4:54:54 PM, you wrote:
rC Alessio, Sergio
So an upgrade is of course necessary.
rC i have upgraded the vigor. Bad news... i am not able
rC to register the draytek anymore. But using a XLite on
rC my pc behind the Vigor works now fine (no one way
Dustin Wildes wrote:
Hello Everyone!
For all of you PhoneCALL users, we have a treat for you today as
PhoneCALL 2.7-RC1 has been released!
Demo with demo/demo doesn't work.
Is that possible to have a look to PhoneCALL without installing it (use
the demo or screenshots)?
Thanks,
Benoît
--
look for intel ambient MD3200 chipset in those modems, there were available here in pakistan a year back but now they are vanished from the market. I've been usingmodem with intel ambient md3200 chipsetwith asterisk and they work fine.
regards,
Umair bari
On 11/24/05, Kunhikrishnan, Salil
Hi,
I had doubt like can asterisk talk ISUP over SS7 which
the normal PSTN
softswitches talk with other switches?
It becomes *necessary* that asterisk *should* talk
with other softswitches
in PSTN using ISUP/SS7 ??
Regards,
Somesh S. Shanbhag
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
good morning!
last week i've started a disa service with approx. 500 calls/day. it is
used by people with cell phones who dial in to asterisk and get the
second tone to dial international pstn numbers(therefore it's zap
only). since disa is the only application in my asterisk installation
used by
I think this series is not available in the market
... Is there any other alternatives ..
The brands which I have mentioned in the
initial post is all winmodems ( or softmodems ). Can we use any of these modem
for this purpose ..
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Somesh S Shanbhag wrote:
Hi,
I had doubt like can asterisk talk ISUP over SS7 which
the normal PSTN
softswitches talk with other switches?
It becomes *necessary* that asterisk *should* talk
with other softswitches
in PSTN using ISUP/SS7 ??
At this point, the standard version of
Title: Aastra 1.3 firmware
Yes I had also noticed this. Also setting dns2 to
0.0.0.0 in the config fileis ignored and I couldn't set the timezone via
the config I had toconfigure it on the phone. Anyone have any other
issues?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
This is how it works by default or you couldn't get a call from a
remote SIP users. The call will drop into the 's' extension (assuming
1.x here - haven't looked at the changes in 1.2 yet) in whatever
context you have configured in sip.conf (default?). The authentication
details are important when
From: Bartosz Wegrzyn - asterisk [EMAIL PROTECTED]
Hi,
After I compile asterisk v.1.2 is tells me that last thing to do is to
make install. Unfortunately it goes it to loop after I type make
install
this is the loop:
else \
mv include/asterisk/version.h.tmp
Wayne Gemmell wrote:
I'm having trouble receiving faxes using rxfax. Could somebody please browse
my log file and give me a swift kick in the right direction? I've also added
my zapata.conf file at the end.
have you tried using direct indialing, to see if rxfax works? (I assume
you are now
On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.
Making outbound calls does result in Error 400 - exept if I do call my
own phonenumber.
I dind find the solution th this problem in current CVS source,
chan_sip.c has to be updated.
Elmar
i enabled hint for
some of my SIP and SCCP lines but on all i have Temp Fail and unavailable line
status.
when i make: SIP
SHOW SUBSCRIPTIONS nothing is shown
call-limit =
2useclientcode=yesnotifyringing=yes
is in the
config.
Somebody can help
me?
Hi Avi,
I added a bit the Asterisk wiki to explain hopefully more clearly how to get it
installed.
Please have a look at:
http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN
Any feedback appreciated.
After you have installed the Diva Server drivers, please install the
chan_capi_cm
Hi, this morning I've switched our 'in-line' Asterisk system (between
legacy PBX and PSTN) live after a few false starts with the PBX
configuration.
I've been executing 'show channels' probably hundreds of times, and I
wanted to see show channel Zap/64-1 So I start with 'show cha' and
David Waugh wrote:
I added a bit the Asterisk wiki to explain hopefully more clearly how to get it
installed.
Great, thanks. Once I have a working /etc/asterisk/capi.conf for the
V-4BRI, I'll be sure to add that to the Wiki page for future reference.
cYa,
Avi
--
National Manager - Special
Hello again.
With the Diva Server 4BRI card - remember that these are in effect 4 CAPI
controllers. Eg CAPI controller 1, controller 2, Controller 3 and Controller 4.
Therefore you should have 4 sections in your CAPI conf as follows.
; CAPI config
;
;
general
nationalprefix=0
On Thu, 2005-11-24 at 10:09 +0100, Olle E. Johansson wrote:
Somesh S Shanbhag wrote:
Hi,
I had doubt like can asterisk talk ISUP over SS7 which
the normal PSTN
softswitches talk with other switches?
It becomes *necessary* that asterisk *should* talk
with other softswitches
in
I know that's a real newbie question, but I have a problem.
I keep getting frame rejects, and a D-channel bouncing up and down. BT
say that it is at my end. If I stop asterisk, stop the zaptel service
and restart, things seem ok for a while.
I posted a similar problem a couple of days ago,
On Thursday 24 November 2005 11:37, Kristof Hardy wrote:
have you tried using direct indialing, to see if rxfax works? (I assume
you are now using fax-detection) That way we know if the detection is
failing or the receiving itself.. (or both :))
I'm not sure what you mean, are you saying that I
I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL: 1. make menuconfig 2. make dep 3. ./install.sh 4. copy the file zapata.cong and zaptel.conf 5. modprobe zaptel 6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I
On Thu, Nov 24, 2005 at 12:00:18AM +0100, Fred Blaise wrote:
On Wed, 2005-11-23 at 19:58 +0100, Fred Blaise wrote:
Hi all
I am having an issue when trying to 'make astman'. Asterisk 1.2.0 here,
from source, on debian sarge. Everything else working fine (only SIP
setup anyway)
On Thu, Nov 24, 2005 at 03:16:45PM +0800, Dulmandakh Sukhbaatar wrote:
Tzafrir Cohen wrote:
On Tue, Nov 22, 2005 at 02:00:11PM -0700, Matt wrote:
Looks like you need to install the kernel headers package. While you
are at it be sure that you have the kernel source package installed
On Wed, 2005-11-23 at 08:30 -0700, [EMAIL PROTECTED] wrote:
Does anyone know of a brute force that will work on a serial interface like
hyperterminal?
Look at expect... you should be able to throw something simple together
using a shell + expect script...
ie, connect and
expect login:
send
Wayne Gemmell wrote:
I'm not sure what you mean, are you saying that I should some how circumvent
the menu system to make calls go directly to the fax? Then I should listen
for noises?
Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
witht the parameters)
At least you
On Wed, 2005-11-23 at 12:53 -0800, Anthony Rodgers wrote:
Hi Dave,
exten = callpark,1,Dial(SIP/1000) didn't work - invalid extension
What about:
exten = callpark,1,Dial(Local/[EMAIL PROTECTED])
Regards,
Adam
___
--Bandwidth and Colocation
asterisk183 wrote:
5. modprobe zaptel
6.
But when I doing insmod qozap.o
and ztcfg don't start because in /qozap directory I don't have qozap.o
files. Why?
what is the output you get 'after' you do: modprobe qozap
After this, what is the output you get after: ztcfg -v
Also, the qozap.ko
On Thursday 24 November 2005 12:49, Kristof Hardy wrote:
Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
witht the parameters)
At least you should hear a 'fax' answering.
Thanks, I'll try that.
Hm, you could try enabling the busy detection in your zapata file..
As of
Title: Fax sending problems
Hi, I've got iaxmodem setup but I'm getting failed fax sending. When I send a fax it is spooled through the system and I hear the destination fax machine pick up, it's sat near me, and the transfer starts. However after about 30 seconds the line drops and the fax
On Thu, Nov 24, 2005 at 11:28:00AM +0100, asterisk183 wrote:
I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the
instruction in INSTALL:
1. make menuconfig
2. make dep
3. ./install.sh
4. copy the file zapata.cong and zaptel.conf
5. modprobe zaptel
6.
But
Show: insmod: qozap.o: No such file or directoryKristof Hardy [EMAIL PROTECTED] ha scritto: asterisk183 wrote: 5. modprobe zaptel 6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why?what is the output you get 'after' you
On Thu, 2005-11-24 at 10:26 +, Julian Lyndon-Smith wrote:
I posted a similar problem a couple of days ago, and one of the
responses suggested that the TE4xxP may be on it's way out.
Is there any way of testing this card to see if that may be the case ?
Speak to digium and ask them how to
kernel 2.4 Tzafrir Cohen [EMAIL PROTECTED] ha scritto: On Thu, Nov 24, 2005 at 11:28:00AM +0100, asterisk183 wrote: I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL: 1. make menuconfig 2. make dep 3. ./install.sh 4. copy the file zapata.cong and
asterisk183 wrote:
and ztcfg don't start because in /qozap directory I don't have
qozap.o files. Why?
If you don't have qozap.o files, then your qozap is not compiled
correctly. Try (in qozap dir) a 'make clean' and 'make all' and see if
this produces an error.
Hello friends,
I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I
have three SIP phones and one H323 phones connected to asterisk. The problem is
that when I dial an invalid extension from H323 phones, I get the invalid
extension message with exten = i... in that
What firmware version did you use for the polycom phone ??
I just tried it on my IP600, and when I press the park button, it waits
for me to dial an extension number, then I press park again, and it just
hangs up the call.
Thanks,
Adam
On Tue, 2005-11-22 at 13:56 -0800, Anthony Rodgers wrote:
Bom dia,
Gostaria de saber se alguém já conseguiu utilizar o pabx intelbras como
conexão pstn para o asterisk?!?!
Já li algumas coisas sobre alterar o wcfxs.c mas nada que tenha surtido
efeito.
Agradeço a atenção.
Att,
Thiago Rodrigues
--
#
# THIAGO
I have had my linesman go over the lines on the pole and manhole and
remake the connections, I have played with the milliwatt generator to
dial out and back to another number, measuring with ztmonitor to
establish the levels, and I have played with the echo canceller
settings. I still get hum
Does asterisk have support for SIP session timers?
David
On 11/24/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
Matt Riddell wrote:
Kevin P. Fleming wrote:
Matt Riddell wrote:
So how does Asterisk know that the media stream has been disconnected
between
the two remote hosts?
It
written by David Waugh
Hi Avi,
I added a bit the Asterisk wiki to explain hopefully more clearly how to
get it installed.
Please have a look at:
http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN
Any feedback appreciated.
After you have installed the Diva Server drivers, please
Erik Slooff ha scritto:
I would like to suggest one small addition for clarity:
you will *need* to have isdn4linux and capi4linux installed on your system
in order to get chan_capi-cm installed.
You just need the capi20 lib in order to use the chan_capi
wget
I would like to suggest one small addition for clarity:
you will *need* to have isdn4linux and capi4linux installed on your
system
in order to get chan_capi-cm installed.
You just need the capi20 lib in order to use the chan_capi
wget
[EMAIL PROTECTED] wrote:
Hello everybody :-)
This are my first line french zapata.conf settings.
I have 3 like this, with only rx/tx gain a little bit different levels.
Running well.
Best Regards,
Francois BERGERET,
France.
usecallerid=yes
hidecallerid=no
usecallingpres=yes
Ok, pay attention to /dev/capi20 device it must exists with the right
permissions
You just need the capi20 lib in order to use the chan_capi
wget
ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2
tar xvjf isdn4k*bz2
cd isdn4*
./configure
make
make
Hi Erik,
You'll have to excuse my ignorance here. But why is this?
I don't have isdn4linux and capi4linux installed but do have
isdn4k-utils-devel-3.2-13.p1.1
isdn4k-utils-3.2-13.p1.1
installed.
Is this for the capi20.h needed for chan_capi to compile?
thanks David
hi,
how can i hangup such calls without restarting asterisk?
the Zap channel on this case is busy for more than 7 hours
some logs are followed.
thanks,
Paradise Dove
-
Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call
'SIP/2378-740f' for lack of RTP activity in 25788
David Waugh ha scritto:
I don't have isdn4linux and capi4linux installed but do have
isdn4k-utils-devel-3.2-13.p1.1
isdn4k-utils-3.2-13.p1.1
Those are old packages, I suggest you to uninstall it and manual compile
the version I posted in a previous release
There are alot of changes in
Hello,
Is there a GUI to manage sip users and voicemail with
Asterisk Realtime .
Regards
Harry
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez
Hi people,
I just rolled out my first attempt with Asterisk to get a working PBX.
I'm recieving my calls still through my ISDN connection. But I'm
getting a lag of almost 1 second between the both sides of the
conversation. What should be the first things to look at when trying
to solve this lag?
Did you get it? I would like to take a whack at it if not.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 23, 2005 10:30 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Help need to
On Tue, 22 Nov 2005, Lenz wrote:
I also have never found anybody running an Asterisk system using app_icd.
Maybe app_queue is now after all flexible enough to be used in most cases.
Anybody else using different apps for Asterisk call centre applications?
I suspect that since the authors
Benoît Mérouze wrote:
Dustin Wildes wrote:
Hello Everyone!
For all of you PhoneCALL users, we have a treat for you today as
PhoneCALL 2.7-RC1 has been released!
Demo with demo/demo doesn't work.
Is that possible to have a look to PhoneCALL without installing it
(use the demo or
On Mon, 21 Nov 2005, Bob Knight wrote:
Just pulled a v1-2 onto a system that was running a v1-0.
Zaptel and libpri, build and install just fine.
Building asterisk is fine.
But when I try to do a make install on asterisk, it goes into an
infinite loop doing on .depend doing:
I had the same problem with ISDN. I actually got the last second or so of the previous call played back at the beginning of each call. There is a patch for this problem. I wish I could remember the name of the person who sent it to me. Maybe she will contact you if she sees this post. I will also
On Mon, 21 Nov 2005, Jonathan k. Creasy wrote:
I've thought about doing that as I have a few spare also. I would use
the raq4 I think.
Let me know if you have any trouble with it.
What you may want to do (I have several of these) is see if you can
re-install the new Centos + BlueQuartz
I found running a later kernel and source code fixed it. I had it on
Fedora Core 3 using kernel 2.6.9 but after updating to 2.6.12 the
problem went away.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: 24 November 2005 14:16
Doug Lytle wrote:
They must have fixed it, because I just logged in. Looks nice, will
have to give it a try this long holiday weekend.
Doug
Hey Doug - yes, it was fixed this morning - we'd purged all the old demo
data forgot to re-create the demo account.
We've already gotten quite a few
I'm all for criticism where it's due but I'm sure for all the bashing of
Voipjet going on in this thread I'm sure there are just as many
non-users who are generally happy with the service they provide and
the price at which they provide it.
I for one am also a customer of Verizon, a fact I'd
I have a QuadBRI card installed, and I received the call incoming, but I don't place call outgoing. Asterisk show this message: Executing Dial("SIP/101-a440", "ZAP/g1/3472543320|60") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3472543320 Nov 24 15:43:14
On Thursday 24 November 2005 12:49, Kristof Hardy wrote:
Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
witht the parameters)
At least you should hear a 'fax' answering.
Yes, I hear a fax answering, so at least I know its working.
--
Regards
Wayne Gemmell
Work:
Title: jittering with Iax2 and Meetme on Asterisk 1.2.0
Hi
I have been using Asterisk 1.0.9 fairly successfully. I have Iax2 softphones based on the IaxClient library that are dialing into Meetme conferences. I am using a Zaptel card as a timing source.
I am now trying to migrate to
David Thomas wrote:
Does asterisk have support for SIP session timers?
No.
/O
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello,
I have compiled asterisk cvs under freebsd, no problems.
When starting asterisk, I get :
[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined symbol ast_config_load
What's wrong?
Olivier
Hello,
Read the Makefile in apps.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Hello,
I have compiled asterisk cvs under freebsd, no
problems.
When starting asterisk, I get :
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
/libexec/ld-elf.so.1:
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:
[snip]
Well, as the user stated on the original message, the asterisk
server is behind a NAT and the client is also behind a NAT….
if you make it work just by opening ports, let me know..I have
never been able to get it to work, that’s why
I have a basic system working, except for callerid. The Polycom 500 just
shows call from Business Line on the screen. Business Line is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a
Wolfgang S. Rupprecht wrote:
Klaus Darilion [EMAIL PROTECTED] writes:
There is a new ietf WG to come which deals with peering issues. It's
called SPEER (formerly VOIPEER)
The list archive is at
http://darkwing.uoregon.edu/~llynch/voipeer/
minutes from last ietf meeting:
This is my network scheme:
h323 endpoint1 ... endpint10 = gk1 = gk2 = asterisk
GK1 configuration: routed mode
GK2 configuration: direct mode
How to obtain that rtp channels not through asterisk for h323 to h323 calls
Thanks in advance!
___
Tes réponses sont aussi sybillines que tes questions :)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 16:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users]
Hi,
I have a problem with e-mail notifications. For some reason Asterisk
does not use the serveremail configuration when sending e-mails
notifications. it always send it using [EMAIL PROTECTED]
My configuration:
pbxskip=yes ; Don't put [PBX]: in the subject line
[EMAIL
Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Tes réponses sont aussi sybillines que tes questions
:)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part de harry gaillac
Envoyé :
Happy Thanksgiving everyone.. I added the following page to the Wiki
documenting how I solved this problem without having to hack with ICD or
any commercial offerings.
http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback
Hope it can help somebody out.
tf.
-Forwarded Message-
On Thu, 2005-11-24 at 17:54 +0100, harry gaillac wrote:
Je ne connais pas la signification de sybillines.
http://www.village-justice.com/forum/viewtopic.php?t=1224start=0sid=964c2c9a1cd842eaca284be8899028a8
--
Dave Cotton [EMAIL PROTECTED]
___
I found out that I have a faulty Belkin Router which was causing the
problem. I tried forwarding ports as well as DMZ'd the Sip device but
still could'nt not hear the voice. So i plugged the sip device directly
to the cable modem it worked fine. So my guess is that though I
have set up the router
This is usually a problem one of the pair not physically
disconnected, i.e loose plug/socket connection. Try
to replace your connectors and test your cable
David
Yat Sin
Sangoma
Technologies
(905) 474-1990
x119
(800)
388-2475 x119
Fax:
(905) 474 9223
MSN:
[EMAIL PROTECTED]
Email:
Walter Willis wrote:
not work fine
Actually it is recognized as an x100p device:
Nov 21 19:54:34 asterix kernel: Zapata Telephony Interface Registered on
major 196
Nov 21 19:54:34 asterix kernel: PCI: Found IRQ 5 for device 00:0d.0
Nov 21 19:54:34 asterix kernel: wcfxo: DAA mode is 'FCC'
SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il n'est guère usité au
sens propre que dans ces locutions : Les oracles, les livres, les vers
sibyllins, Les oracles, les livres, les vers des sibylles.
Il signifie au figuré Qui est mystérieux obscur, dont le sens est difficile
à saisir. Il m'a
I found the mail from Pauline Middelink!
filename: hfc_pci.c.diff
--- /root/hfc_pci.c Wed Aug 7 15:31:24 2002
+++ /usr/src/linux/drivers/isdn/hisax/hfc_pci.c Thu Oct 31 10:18:05 2002
@@ -270,8 +270,16 @@
if (fifo_state)
cs-hw.hfcpci.fifo_en ^= fifo_state;
Tom Vile wrote:
Everyone,
I have a TDM400 REV I Ver 1 board and am having an issue with 1 of the
4 FXO channels. FXO 1 always has clicks, pops and echo but the others
are crystal clear all of the time. The card is on its own IRQ zztest
shows 100% to 99.98% and is getting 1000 int per
Cela veut simplement dire que tu te plains de ne pas avoir de réponses, mais
qu'en fait tu n'en donnes pas non plus, sauf sous forme de devinette.
Auquel cas, il est plus simple de ne pas répondre,
merci
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Hi,
Anyone has experiences with sending faxes using Asterisk and a TE405P
Digium card (or similar PRI) with a PRI connection?
Any insights wanted, bood, bad and ugly.
Thanks,
Andre
___
--Bandwidth and Colocation sponsored by Easynews.com --
Klaus Darilion [EMAIL PROTECTED] writes:
It's not that easy. If you want to have open SIP URIs (just like email
is open for everybody) you will receive SPIT calls. E.g. the SPEER
group tries to define rules for VoIP peering which allows
authentication to enable open SIP URIs. (I won't open
Check your logs, make sure you are waiting long enough before sending
the call to the polycom.
Uf asterisk sees the CID, it should send it and it should show up on the
polycom.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
MacKay
Sent:
I do believe there is a system reset is there not? Thought I saw it in
the manual.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, November 24, 2005 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Nov 23, 2005, at 1:10 PM, Denis Vella wrote:
Hi,
I'm trying to use modems with Asterisk+VoIP Gateways in an attempt at providing an Internet service.
Home_PC-->Modem-->PSTN-->VoIP_Gateway_FXO-->Ethernet-->Asterisk-->Ethernet-->VoIP_Gateway_FXS-->Modem-->PPP_Server-->Internet
I've been
Hi there,
for testing purposes I am searching for a freely available softphone that
supports SIP subscriptions and display the status of a few of these via
e.g. a simulated LED. I know about
* EyeBeam (not free)
* SNOM softphone (needs Win XP and has old firmware)
Are there other softphones
UTStarCom has the F3000 coming in December, which will have according
to their spec
* WEP (64 and 128 bit )/WPA/MD5 Auth
* Handover/Roaming between different AP and SSID
So what else is different compared to the F1000? The 1000 also does
WEP 64/128 and WPA with the newest firmware.
i have the 1.6.3 firmware and also when i press park i need to dial another extension..
On 11/24/05, Adam Goryachev [EMAIL PROTECTED] wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension
Hi... I have the polycom 301 with firmware 1.6.3
When i Press Park, i get a dialog to enter a extension.
A dial 700 ther
and the call get parked, and i recive a call announceme where the calls was parked.
is this normal ???
On 11/24/05, Alvaro Parres [EMAIL PROTECTED] wrote:
i have the 1.6.3
On Wed, November 23, 2005 20:29, Francesco Peeters said:
On Wed, November 23, 2005 11:17, Francesco Peeters said:
SNIP
Just a question: Does the card require a device connected to it to start
up in NT mode? I have been testing so far, so I have not yet connected my
phone to the card with a
Kevin Ragsdale wrote:
Has anyone tried the newest Polycom firmware? The release notes
indicate they have added support for a new BLA draft.
TIA,
Kevin
Does anyone know if this new firmware support watching more than 7
buddies at a time?
Cheers,
Kevin
--
Optimacy Communications, LLC
I use putty.exe it works wonders.
available here:
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
You need ssh running on linux for it to work.
On 11/24/05, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote:
Hi,
Does anyone know of a Asterisk Manager Interface
Hi,
I have compiled chan_bluetooth on FC4 (kernel 2.6.14-1).
The phone (SonyEricsson W800i) is paired with the BT dongle (ID
0db0:1967 Micro Star International Bluetooth Dongle).
I have configured vi /etc/asterisk/bluetooth.conf like that:
[general]
rfchannel_hs = 2
rfchannel_ag = 3
According to this not:
http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,5082,00.pdf
but they do mentions some new blf support, so go figure.
On 11/24/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Kevin Ragsdale wrote:
Has anyone tried the newest Polycom firmware? The release notes
http://ipswitchboard.thorben.dk/index.php?option=com_contenttask=viewid=26
Itemid=46
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: den 24 november 2005 20:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Without putty, my windows would be meaningless.
PaulH
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 25, 2005 6:21 AM
Subject: Re: [Asterisk-Users] Looking for Windows
Hi Erik,
You'll have to excuse my ignorance here. But why is this?
I don't have isdn4linux and capi4linux installed but do have
isdn4k-utils-devel-3.2-13.p1.1
isdn4k-utils-3.2-13.p1.1
installed.
Is this for the capi20.h needed for chan_capi to compile?
thanks David
Hi Dave,
On
Excellent work!
PaulH
- Original Message -
From: Tyler [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, November 25, 2005 4:05 AM
Subject: [Asterisk-Users] Re: Queue Callback - SOLVED
Happy Thanksgiving everyone.. I added the following page to the Wiki
Hi Adam,
Same - the parkee gets the stall number announcement instead of the
parker.
On Nov 24, 2005, at 2:49 AM, Adam Goryachev wrote:
On Wed, 2005-11-23 at 12:53 -0800, Anthony Rodgers wrote:
Hi Dave,
exten = callpark,1,Dial(SIP/1000) didn't work - invalid extension
What about:
exten
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