Yup, I found their support very unhelpful and unwilling to go the
extra (or even the first) mile..
On Sat, 15 Jan 2005 18:27:49 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Eric Bishop wrote:
I logged a support issue with HP and their response was that it's not
their server that is the
Bilal Ghayad wrote:
Dear Sebastian;
Thanks a lot for your kindly advise to use ASTCC.
But can u advise me the link for ASTCC to download it and wether it is open
source (to download the source and work on it?
Regards
Bilal
_
check it out
http://www.voip-info.org/wiki-ASTCC
regards
I have researched my issue a little more and this is what I have come up
with. Here a examples of my configurations so far and the error I get when
I try to dial an external number. It seems like I am so close, thanks for
the help so far!
Chris
Steve,
Highly interested as well, let me know with the other interested parties.
Hadi.
-Original Message-
From: Ben Merrills [mailto:[EMAIL PROTECTED]
Sent: 14 January 2005 09:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SS7 and
Has anyone tried SunRocket with Asterisk?
http://www.sunrocket.com/
The $199/yr. plan seems like an excellent value,
and most reviews have been favorable.
However, I don't know if it is possible to obtain the SIP
credentials, so one can bypass their gizmo.
Thanks,
Stewart
hi list,
im using xlite softphone clients in the windows box, and i can make a
call (connection) in each other, but im wondering why is that i cannot
hear any voice from client's end (vice-versa), i already checkd the
volume of the speaker, is there something missing in the configuration
file
i cannot
hear any voice from client's end
This is a common problem if you are using NAT (behind router)
Google for asterisk one way audio
and take a loog here
http://www.voip-info.org/wiki-Asterisk+SIP+NAT
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Can some one advise me an PC to Phone client software to be used under
Windows OS at the client side, to be communicated with Asterisk PBX?
Phones hard and soft
http://www.voip-info.org/wiki-VOIP+Phones
A few clients I've tried
for SIP X-Lite, SJPhone
for IAX2 IAXPhone, Firefly, IAXComm,
Hi, can't get X100P (fully zapata compatible clone) to work (I'm in
Australia).
* recognises the card and the channel (1) but has definetely some problems
talking to the pots line.
I set up this simple dialplan for ZAP (incoming context, as setup in
zapata.conf, for channel 1)
[incoming]
exten
Ok, here's a weird one.
I've attached a spa2000 to asterisk, and got the two phones to register
as exten 706 and 707.
I can call exten 708 (a cisco 7940) from 707 and everything works fine.
I can call exten 708 from 706 and everything works fine.
When I make a call to either 706 or 707 from any
When I make a call to either 706 or 707 from any phone, the phone
attached to the spa does not ring. However, if I pick up the appropriate
phone, the connection is made and normal conversation can take place.
I had the same problem with a Cisco 827-4V. It turned out that the
phones were fussy
Is there even a seperate LED under the message button? I'd doubt it
but I've not had
any of my BT's in bits yet!
Whilst on the subject of BT's, do the callers and called buttons function?
they dont seem to do anything on mine?
thanks
Mike
On Fri, 14 Jan 2005 23:47:12 -0700, Paul Fielding
[EMAIL PROTECTED] is believed to have said:
Budgetone and MWI
The message button can be programmed to dial an extension that checks
voicemail
exten = 160,1,Voicemailmain(${CALLERIDNUM})
Thanks, this is what I was thinking about. Still, how do you get the BT
to dial 160?
In my Asterisk
Does echotraining also work with HFC-S controllers or is this feature
restricted to some digium controllers?
So does it make sense at all to switch it on for HFC-S controller used
in NT-Mode with ISDN-Phones connected?
Jui
Rich Adamson wrote:
I have echo training set on in my zapata.conf file
Hi , I tried that and asterisk console says:
Jan 15 11:11:13 WARNING[3637266]: pbx.c:1922 ast_pbx_run: Invalid
extension '*', but no rule 'i' in context 'inbound-line2'
-- Hungup 'Zap/1-1'
my context says:
[inbound-line2]
Exten = s,1,Answer
Exten = s,2,SetMusicOnHold(slimp3)
Exten =
In article [EMAIL PROTECTED],
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Matthew Boehm wrote:
So are you telling me that you cannot use other commercial products in
conjunction with asterisk?
You cannot distribute a closed source add-on (except AGI) for Asterisk
without a
Oops, I replaced the a's below with *'s and it works now.
Mike
On Sat, 15 Jan 2005 11:21:42 +, Mike Dent [EMAIL PROTECTED] wrote:
Hi , I tried that and asterisk console says:
Jan 15 11:11:13 WARNING[3637266]: pbx.c:1922 ast_pbx_run: Invalid
extension '*', but no rule 'i' in context
[EMAIL PROTECTED] is believed to have said:
Other than doing an IVR type arrangement or a phone number dedicated to
VM access is there a way to do this? On my old POTS line I used to be
able to call my line and simply punch * during unavailable message
playback to go to the equivalent of
On 15 Jan 2005, at 11:26, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Matthew Boehm wrote:
So are you telling me that you cannot use other commercial products
in
conjunction with asterisk?
You cannot distribute a closed source add-on
Hello list,
Did someone know where to find the SIP image for
Packet8 DTA 310 box to work with Asterisk??
Thanks.
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Hi all,
Anyone set up * - ipcb.net ?
Feel free to contact me off list for some conf examples, cant seam to
get it right ;)
Regards
/Danny
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Aldo Bergamini is believed to have said:
This is how it was before:
; EXT. 2XXX
; generic dialer
exten = _2XXX,1,Dial(SIP/${EXTEN},20)
exten = _2XXX,2,Voicemail(u${EXTEN})
exten = _2XXX,3,Hangup()
exten = _2XXX,102,Voicemail(b${EXTEN})
exten = _2XXX,103,Hangup()
And this is how I changed
Dear Group,
This might be a little OT, but I am lost in this.
What type of network switches is the minimum for use of an asterisk in a 70
telephones and 68 computers environment.
Can unmanaged switches do the job or is Level 2 or level 3 necessary an why?
The sound quality has to be as good as
hi, thanks for the reply. i just want to clarify that my asterisk box
and client softphone are in the same network as follows:
1. 192.168.17.57 = asterisk
2. 192.168.17.59 = client1
3. 192.168.17.60 = client2
so you mean that the even the same network, the firewall will affect the
channeling of
On Sat, 15 Jan 2005, tim panton wrote:
Erm, at the risk of getting flamed, where does IAX come into this
picture? If I re-implement IAX(2) in a different language (not using
iaxcomm except as a refererence or test ) and want to sell a product
based on it can I do that, or do I need a license
Yup, I found their support very unhelpful and unwilling to go the
extra (or even the first) mile..
Might ACPI (not APIC) have anything to do with this condition? I once had
a hard time with a bunch of cards which were not taking interrupts. I
disabled ACPI interrupt routing (from
Dear all,
all you chan_capi users out there, has anybody a AVM Fritz!AP-ISDN
working with Asterisk?
The sound on the outbound side is very stuttering (not possible to
understand). Muting the remote end did not have any effect while muting
the local end stopped giving me below errors:
tim panton wrote:
Erm, at the risk of getting flamed, where does IAX come into this
picture?
If I re-implement IAX(2) in a different language (not using iaxcomm
except as
a refererence or test )
and want to sell a product based on it can I do that, or do I need a
license ?
(I come from a BSD
Hi,
Just a question,
For you, what is the more reliable kernel for an asterisk prod server...
Thanks
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On January 15, 2005 06:21 am, Mike Dent wrote:
Jan 15 11:11:13 WARNING[3637266]: pbx.c:1922 ast_pbx_run: Invalid
extension '*', but no rule 'i' in context 'inbound-line2'
-- Hungup 'Zap/1-1'
That's because 'a' is used when you're *in* the VoiceMail app... i.e. you
need to be hearing
On January 14, 2005 08:55 pm, Steve Underwood wrote:
If you have the source for spandsp-0.0.6 can you send it to me, please.
I'm only up to 0.0.2pre7 :-)
:-) Sorry, I meant 0.0.2pre6.
-A.
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Hi,
With chan-capi driver you have the command CapiCD to deflect a call on the D-channel (as described on www.junghanns.net)
Does any one have an example (of the dial command) how to do the same thing with the junghanns drivers and the QuadBRI card? Would there be anything additional to
Em Sex 14 Jan 2005 18:03, Bruno Hertz escreveu:
On Fri, 2005-01-14 at 16:27 -0200, Denis Galvo - iSolve wrote:
Em Sex 14 Jan 2005 16:11, Dan escreveu:
I dont have problems when calling PSTN extensions, and calling
VoceMail, EchoTest, etc. The problem is related with the conversation
i have no problem with 2.6.
On Sat, 15 Jan 2005 13:12:18 +, Jeremy SALMON
[EMAIL PROTECTED] wrote:
Hi,
Just a question,
For you, what is the more reliable kernel for an asterisk prod server...
Thanks
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Mike Dent wrote:
Whilst on the subject of BT's, do the callers and called buttons function?
they dont seem to do anything on mine?
Yes, but the hand set needs to be off hook.
Doug
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Hi ALL;
I have problem with Proxy-auth. Ihave an
asteriskas a proxy and cisco ATA as UA.
Can anybody help??
Regards
Mohammad
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Anyone else ever have the problem of asterisk picking up with voicemail
after one ring on an extension? I'm using free world dial up's IAX2 service,
and I can make calls but received calls get a voicemail pickup after one
ring. No decent answer on google, cannot find anything that seems to be
Don't know, but certainly easy enough for you to try.
Does echotraining also work with HFC-S controllers or is this feature
restricted to some digium controllers?
So does it make sense at all to switch it on for HFC-S controller used
in NT-Mode with ISDN-Phones
One thing that jumps out at me is your Dial line:
exten = _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
I believe this should be:
exten = _9X.,1,Dial(SIP/voicepulse-out/${EXTEN:1},30,r)
What I think is happening is the ${EXTEN:1}@ is being treated as the
username when contacting voicepulse,
Rushowr wrote:
Anyone else ever have the problem of asterisk picking up with voicemail
after one ring on an extension? I'm using free world dial up's IAX2 service,
and I can make calls but received calls get a voicemail pickup after one
ring. No decent answer on google, cannot find anything that
This link might help:
http://www.dslreports.com/forum/remark,11775216~mode=flat
On Fri, 14 Jan 2005 23:29:34 -0500, Randy [EMAIL PROTECTED] wrote:
Chris,
I do not have VoicePulse Open Access, but I do have an incoming number through
VoicePulse Connect. You might want to give the following
Can you show us the CLI output of what is happening?
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Hi,
I currently have Asterisk running behind a linux router running nat.
Clients register with the public address and when the sip requests
reach the router, port forwarding is used to divert the traffic to *
i.e. all sip and rtp go to the asterisk box.
I now want to set up ser (so that i can
CLI output is (currently unavailable as it only happens when another PERSON
calls me, the call me script from FWD works fine).CLI Output is something
similar to:
IAX2/550949/7 is ringing
IAX2/550949/7 answered IAX2/[EMAIL PROTECTED]:4569/6
Attempting native bridge of
Rushowr wrote:
But then the CLI puts out a message concerning congestion and kicks the call
directly to voicemail.
The problem is that the DIAL fails. You don't have a voicemail problem.
You have a problem with the peer you are trying to dial. The dial
fails, Asterisk considers it a
Hello I have asterisk connected to acisco gateway and a broadsoft softswitch. My SIP phones are custom software on a PC. I make a call via the gateway into asterisk and a 2-way talk path is setup from the gateway to the sip PC. Next the PC (PC1) uses a SIP refer to transfer the call to the
What is ${FWDRINGS} set to?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rushowr
Sent: January 15, 2005 11:13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail after one ring?
CLI output is
Jonathan,
Log on to [EMAIL PROTECTED] with user name :root password: password
Then run: netconfig
You'll probably find your nic wasn't automatically assigned an IP
address.
You will need to enter in all of the details manually.
Cheers,
Dean
-Original Message-
From: [EMAIL
so you mean that the even the same network, the firewall will affect the
channeling of data accross the asterisk?
Not likely. Look for the transmit silence setting on X-Lite which is
another gotcha that causes no audio and or try other SIP (and why not
IAX) clients.
Program your locked ADSI phone not by unlocking it but by using the
feature slot FDN download descriptors of ADSI SEC security code telco
provider lock number key from database list. ch33s3
First reset phone. Set clock to Jan 1 12:00AM (options-2?). exit to
main screen. press options,
Hi friends
I recently bought an X100P card and fixed it to my
PC.
I connected one port to an analog phone and the
other(the port which is supposed to be connected to
the telephone network) is left unconnected to anyone.
When tried to make a call from asterisk command prompt
to the phone I am
Mohammed,
If you tell us what the problem is we may be able to...
I have problem with Proxy-auth. I have an asterisk as a proxy and
cisco ATA as UA.
Can anybody help??
Describe the problem, and give any information which is shown on the
asterisk console.
Phil.
Hi..
I need to send calls
coming from SIP phones behind asterisk to Cisco Call Manager 3.3.4. We have
created an H323 trunk on the call manager.Provided that Asterisk-oh323 is
installed, how should h323.conf be configured for that?
Also when this is
done can I setup CCM to alert phones
why linux/moduleparam.h is missing in the source? I
saw it in 2.6 source.
anybody can help?
Thanks!
[EMAIL PROTECTED] zaptel]# make
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE
-DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
Hello list,
I have been running Asterisk CVS for a good while. When I try to
install 1.0.3, asterisk won't start. Below are the last few lines of
output before Asterisk crashes. I ran make samples to start with a
fresh setup.
[app_read.so] = (Read Variable Application)
== Registered
I am trying to
compile oh323 and having the following error. Can anyone help please?! This is
my third post. These are the versions I am using:
Compilation
Error:
--
g++ -o obj_linux_x86_r/simph323
-s -L/root/pwlib/lib -L/root/openh323/lib
Hi Michael,
I just got my TDM04B and I have successfully installed it into an IBM
Netfinity 3000 with a PIII 500 Mhz processor. I'm running Gentoo
2004.3 and haven't had any issues with interrupts.
Good luck!
Calvin
On Jan 13, 2005, at 11:10 AM, Michael Swan wrote:
Hi,
A week or so ago I wrote
Update your CVS
Xu, Duo wrote:
why linux/moduleparam.h is missing in the source? I
saw it in 2.6 source.
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Thor Atle Rustad wrote:
I have been running Asterisk CVS for a good while. When I try to
install 1.0.3, asterisk won't start. Below are the last few lines of
output before Asterisk crashes. I ran make samples to start with a
fresh setup.
[app_realtime.so]Jan 15 17:42:24 WARNING[19841]:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of VenkataRao Chimata
Sent: Saturday, January 15, 2005 12:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi friends
I
I have asterisk
CVS-v1-0-12
Can someone please
advise what is the best solution and versions for adding h323 support to
asterisk. I am confused between oh323/pwlib/asteris-oh323 versions. Asterisk
oh323 0.7 README say I need to getPWlib (v1.6.6) and OpenH323 (v1.13.5) but I cannot
find
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of VenkataRao Chimata
Sent: Saturday, January 15, 2005 12:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
Hi friends
I
On Jan 13, 2005, at 11:10 AM, Michael Swan wrote:
A week or so ago I wrote about the problems I was having using a Digium
TDM04B card in a Dell PowerEdge 750 IU running Fedora Core 1. Digium
steadfastly indicates their card won't work in any PowerEdge 650, 700
and 750 series machines do (sic)
you might also want to make sure the X100P is not sharing it's IRQ with any
other card.
Lyle
- Original Message -
From: Emanuele Venditti [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, January 15, 2005 2:36 AM
Subject: [Asterisk-Users] No sound with X100P (clone)
Sam,
I suggest patch the makefile in your asterisk channels directory using
channel_makefile.patch and replace chan_unicall.c.
Kibe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Njenga
Sent: Wednesday, January 12, 2005 1:54 PM
To: Asterisk Users
Jeremy SALMON wrote:
Hi,
Just a question,
For you, what is the more reliable kernel for an asterisk prod
server...
The following 2 recent quotes from kernel developers may be worth
considering when making your decision:
After 2.6.9-ac its clear that the long 2.6.9 process worked very
I have never had an issue with 2.6.9 with asterisk.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Richard Scobie
Sent: Saturday, January 15, 2005 2:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Return
Hi all,
I'm using VOIPJET to make international calls with an IAX2 connection
between my local asterisk server and their server(s).
At times I seem to have a problem if 5 or more international calls are made
at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the
asterisk
Is there any benefit of increasing the registration period of a SIP
device ? I've seen periods of between 120 and 3600, and wondered why.
Julian
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On Sat, 15 Jan 2005 19:58:15 -, Derek Conniffe [EMAIL PROTECTED] wrote:
I have read many times that IAX2 has a signaling overhead (30 or 40kbps?)
and then every channel uses the codec bandwidth.
This is true if you are using IAX2 trunking. This can be enabled with
trunk=yes in your peer
On Sat, 15 Jan 2005 19:58:15 -, Derek Conniffe wrote:
Hi all,
I'm using VOIPJET to make international calls with an IAX2 connection
between my local asterisk server and their server(s).
At times I seem to have a problem if 5 or more international calls are made
at once - I'm on a 1024kbps
Leif Madsen wrote:
This is true if you are using IAX2 trunking. This can be enabled with
trunk=yes in your peer configuration. The other end must also
support the trunking as well.
In addition, the codec you use must support trunking as well. As far as
I can remember (haven't checked in a
Hello,
I have a CAC Channel Bank I with FXS cards. I've the system up and running,
with just 1 issue.
When I make an inbound call, Asterisk says Zap/26 is ringing, however, the
phone never rings. No
lights are lit on the CAC during the calll.
Outbound call works no problem, and the CAC
Hey, whenever I try to load, I get these errors
Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to
bind to 0.0.0.0 port 4569: Address already in use
Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource:
chan_iax2.so: load_module failed, returning -1
== Manager
Have a close listen to digits/h-15 and digits/h-16.
To my ears the latter could be mistaken for the former ... or perhaps I
am more deaf than I think.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just
Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to
bind to 0.0.0.0 port 4569: Address already in use
That is because you already have something listening on the 4569 port.
I think another instance of Asterisk is already running
___
-Original Message-
From: Satchid [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 15, 2005 7:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] switches
Dear Group,
This might be a little OT, but I am lost in this.
What type of network switches is the minimum for use
Who else makes Analog Telephone Adapters with IAX protocol besides
Digium?
I've seen Farfon is advertising their unit but I'm not sure if they
shipped it or not.
Why is so hard for others to support this protocol in their adapters?
SIP is totally unsuitable for firewall traversal for security
Hello,
Im using oh323 (0.6.5) module for asterisk
(1.0.2) and Im unable to
Specify the user using type= option. My configuration
is :
[mihalik]
type=user
host=192.168.1.36
context=incoming-h323
but all traffic coming from
192.168.1.36 VoIP phone is not identified
and is
Did someone know where to find the SIP image for Packet8 DTA
310 box to work with Asterisk ??
You need 0x to define the SIP settings and 0x1234 to get it to work.
Both are available in the VoIP forums at DSLReports.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646)
I'm having trouble with both my Polycom IP600 and
IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a
phone call that lasted 1hr 39mins which is my record so far. I cannot figure out
what is causing the call to terminate and I am hoping somone on this list can
Hello,
I've signed up for a NuFone account, and added the following
instructions to my config files per NufFones directinos:
iax.conf
[NuFone]
type=peer
host=switch-1.nufone.net
secret=password
extensions.conf
(under the [default] context)
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL
Hi, can't get
X100P (fully zapata compatible clone) to work (I'm in Australia).*
recognises the card and the channel (1) but has definetely some problems
talking to the pots line.I set up this simple dialplan for ZAP
("incoming" context, as setup in zapata.conf, for channel
On January 15, 2005 03:11 pm, Leif Madsen wrote:
This is true if you are using IAX2 trunking. This can be enabled with
trunk=yes in your peer configuration. The other end must also
support the trunking as well.
Also if you're using iLBC you need to set the trunking period to 30ms instead
On January 15, 2005 03:39 pm, [EMAIL PROTECTED] wrote:
I have a CAC Channel Bank I with FXS cards. I've the system up and
running, with just 1 issue.
When I make an inbound call, Asterisk says Zap/26 is ringing, however,
the phone never rings. No lights are lit on the CAC during the calll.
Brian,
The other day on IRC I was trying to get someone to listen to me about
this problem, I'm using CVS HEAD 12-29 and * acts up very strange when
I do a xfer to 700 thru astman, I think it can be qualified as a bug.
I think that there should be in the manager.conf a context and maybe
extension
It seems to never fail - after 3 to 5 minutes SIP - IAX calls drop
audio on one side. I place a call out through voipjet, and call
quality is flawless. However a few minutes later the person who I'm
talking to can no longer hear me. I can still hear them.
What should I look for to resolve
On Tue, 11 Jan 2005 21:31:47 -0500, John Bohman [EMAIL PROTECTED] wrote:
I found this to be missing... thank you,
However after restarting, rebooting, and re-building.. no change..
Still no callerid on the Zap channels...
It is however working on my 7960 so I know the feed from the FXO card is
OK, here is the output:
first, when xfering using astman the one being transfered hears the
parking spot (ie 701).
second, if I try doing this using Cisco 7960 that is in a call to SPA
3000 I get the following a million times in the CLI, and
/var/log/asterisk/messages:
Jan 15 20:03:49
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the SayDigits(${CALLERIDNUM}) command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
exten = 34,1,Answer
exten = 34,2,Wait(1)
exten = 34,3,Playback(vm-extension)
exten =
Greg Blakely wrote:
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the SayDigits(${CALLERIDNUM}) command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
CLI show application SendDTMF
-= Info about application 'SendDTMF' =-
I need to demo a Grandstream phone to a customer, but his IT is not able
to give me for the demo a Ethernet cable with DHCP!!!
I need therefore a wireless hub ??? with the upstream connection to
wireless!!!
Since I do not know how to name it I cannot google it!
Has anybody done such thing?
bye
Try getting a 'wireless bridge'
i have one set up with my gradstream sitting next to me... - Netgear
ME101 is the model...
Cheers
Ronald Wiplinger wrote:
I need to demo a Grandstream phone to a customer, but his IT is not
able to give me for the demo a Ethernet cable with DHCP!!!
I need
I need to demo a Grandstream phone to a customer, but his IT is not able
to give me for the demo a Ethernet cable with DHCP!!!
I need therefore a wireless hub ??? with the upstream connection to
wireless!!!
Since I do not know how to name it I cannot google it!
Has anybody done such
I'm monitoring this list in hopes of learning a little about VOIP and
PBX's but I do have experience here - A bridge requires 2 ends so thats
not going to work. Basically what you want to do (as I understand it) is
adapt a wired device to a wireless network - a media adapter.
I've used plenty
Brian West wrote:
I have never had an issue with 2.6.9 with asterisk.
I second that.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
Joseph wrote:
Who else makes Analog Telephone Adapters with IAX protocol besides
Digium?
The PA1688 chip now supports IAX.
There are many phones/ATA units which use this chip.
There is even a page on the wiki about them.
--
Cheers,
Matt Riddell
___
I'm monitoring this list in hopes of learning a little about VOIP and
PBX's but I do have experience here - A bridge requires 2 ends so thats
not going to work. Basically what you want to do (as I understand it) is
adapt a wired device to a wireless network - a media adapter.
I've used plenty
I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to
a standard analog handset with caller id display (US caller ID).
Although it appears that caller id information is coming into asterisk
(it shows up in voicemail), I can not get it to display on the analog
handset.
Is
Do you mean zaptel CVS? I have got the lastest version
of zaptel from digium CVS.
Could you explain it clear to me? I am newbie to this
compilation things.
Thanks!
Is it kernel source problem or zaptel problem. I spent
hours on google to find a clue , but in vain.
Message: 3
Date: Sat, 15 Jan
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