Matthew Boehm wrote:
MailboxExists([EMAIL PROTECTED]): Conditionally branches to priority n+101
if the specified voice mailbox exists
Bingo... MailboxExists([EMAIL PROTECTED]) is more like
MailboxExists([EMAIL PROTECTED]). Omitting the context was my point of
failure. When it's there
Hello, I'm really new on Asterisk.
Is it possible to use a telephone machine connected to a modem as an asterisk
voice input output device? I do not need PSTN connection.
The scheme i'm thinking about is;
user - phone - modem - asterisk - ip - vice versa.
If it is possible can a user dial
Hi,
Is it possible to have an incoming SIP address like
[EMAIL PROTECTED], where sip.mydomain.com points to a box
running Asterisk?
If so, please could someone give an example asterisk config snippet for
this?
If it is possible, I assume ports 5060 and 1-2 need to be opened
in the
Jared Armstrong wrote:
Hi, I have was testing some of the different ring types with my polycom
500, and the ALERT_INFO settings. Now when my phone receives a call it
wont ring.
I had the same thing happen to me - touched the files on the ftp server,
rebooted the phone, it formatted/reinstalled
Hi,
Has anyone got Gossiptel working with Asterisk? - I am having real
problems getting it to register - i'm just getting timeout errors.
Thanks
--ian
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Hello,
I am using Mandrake 10.1.
Howto to install asterisk.
I have downloaded tarball.
I have not installed any hardware yet.
Is it possible to install ?
Thanks
Varun
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On Saturday 04 December 2004 04:43, Nate Carlson wrote:
In other words, if it's something you really want, add more cash to the
bounty, to help encourage the developer to spend more time on it *grin*:
alright, alright - i'll work on it today :-)
~ Theo
--
Theo P. Zourzouvillys
[EMAIL
I am so far unable to get the busy lamps on a Snom 220 to work either with
current cvs or asterisk 1.0.
I am using the hint extension and the Snom 220 just as described in the
mini-howto on:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html
There are also a couple of
HelloEveryone,
I have recently come across these two terms. I am new at Asterisk and do
appreciate your assistance in this. Some tools such as
"astGUIclient" and "Asterisk Management Portal"
require that the phone system be running Zap or
IAX trunks as well as SIP devices. SIP devices are
Hello
Everyone,
I want to start using Asterisk with Cisco IP Phones
7960 / 7940/ and 7905. Any info or help is
appreciated.
I know I'll have to use SIP but if I want to use the phones
off site meaning from my home for example, how can this be
done?
Also, regarding external lines what
On Sat, 4 Dec 2004, Tracy R Reed wrote:
I have created hint priorities in my dialplan:
exten = l00,hint,SIP/100
exten = 100,1,Macro(stdexten,100,SIP/100)
^
I guess it may just be a typo during retyping, but you have 'l' (lower
case L) in the hint line and a '1' (one) in the macro
Hi
For use my isdn card in NT mode I have compiled
chan_misdn.
When I launch asterisk it stop with th e message :
[chan_misdn.so] = (Channel driver for mISDN Support
(Bri/Pri))
== Parsing '/etc/asterisk/misdn.conf': Found
UnLocking config_mutex
== Registered channel type 'mISDN' (This driver
Nguyen Quang Hoa wrote:
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the
On Thu, 2004-12-02 at 16:47, David Filion wrote:
Hi,
Does anybody else have problems like this.
I'm in the UK with a 1mb ADSL service from Eclipse. I have a Draytek
Vigour 2600 ADSL router.
My * box is configured with a public IP address which is presented on
one of the switch ports
Ian Chilton wrote:
I assume ports 5060 and 1-2 need to be opened
in the firewall too.
I don't know much about SIP and firewalls, but opening ten thousand
ports doesn't sound good, you've just knocked 1/6 of your firewall down
:-(
___
Hi Patric;
I interested in your email on "Mon Oct 2004" with
the subject "Howto get voicemail $VM_
vars into externnotify script?".
Have you been able to set up such an application. If yes, plz
help me to find out about that.
Regards
mohammad
Hi,
Is it possible to have an incoming SIP address like
[EMAIL PROTECTED], where sip.mydomain.com points to a box running
Asterisk?
If so, please could someone give an example asterisk config snippet for
this?
If it is possible, I assume ports 5060 and 1-2 need to be opened
in the
Hi,
I assume ports 5060 and 1-2 need to be opened
in the firewall too.
I don't know much about SIP and firewalls, but opening ten thousand
ports doesn't sound good, you've just knocked 1/6 of your firewall down
That's what I thought but I was told it was the only way to get
Hi,
Is it possible to have an incoming SIP address like
[EMAIL PROTECTED], where sip.mydomain.com points to a box running
Asterisk?
If so, please could someone give an example asterisk config snippet
for this?
snip
--ian
Ian, you don't even have to create a subdomain for this.
Include
Hi Martin,
my router is a vanilla 2600, not the V model, as far as I know it has no special
SIP features, other than SIP seeming to crash it when a SIP call is made from
the internet to the * box here! :(
I mentioned the problem on the draytek forum but I;ve not contacted Draytek
themselves per
Title: Message
You
might want to check your phone directory file. In there you can specify a ring
type for a identified incoming caller - perhaps you have specified ring type 0
which is by default silent.
Peter
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hi Shane,
http://www.voip-info.org/wiki-DNS+SRV
http://slacker.com/~nugget/asterisk7.php
The SRV page was useful - i've done that in my domain now.
But, the other page is talking more about dialing sip addresses through
Asterisk rather than incoming sip addresses.
However, after adding the
Thanks! :)
On Sat, 4 Dec 2004 10:19:59 +, Theo P. Zourzouvillys
[EMAIL PROTECTED] wrote:
On Saturday 04 December 2004 04:43, Nate Carlson wrote:
In other words, if it's something you really want, add more cash to the
bounty, to help encourage the developer to spend more time on it
Trying to setup asterisk and zaptel on a Fedora Core 3. Its all working
after reading up on udev but I still get errors.
[EMAIL PROTECTED] ~]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart
Hi all,
I'm debugging a PRI problem, i can see the calling
number but i get a busy all the time. From the output
below, I guess asterisk hangs up immediately. Can
anyone point out what the problem is?
Thanks in advance.
*CLI Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 2
Hi all,
another thing i noticed, when i start asterisk and
type pri show span 1, i get the following:
Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
As soon as i type pri debug
I assume ports 5060 and 1-2 need to be opened
in the firewall too.
I don't know much about SIP and firewalls, but opening ten thousand
ports doesn't sound good, you've just knocked 1/6 of your firewall down
That's what I thought but I was told it was the only way to get
On Friday 29 October 2004 21:17, lenz wrote:
Hello list,
I was looking for a way to implement non-blind call transfers with *, i.e.
the usual behaviour of most PBXs when pressing the flash button:
- A and B are talking
- A pushes flash
- A is free to compose a new number
- B hears music on
Inline...
I am preparing to roll out Asterisk setup with TDM400P, 4 FXO modules in
a small office. Asterisk will replace legacy system (4 telco lines, 8
extensions PBX), but before the new system and ip phones would be
installed, the legacy system is still in use. The four telco lines are
On December 3, 2004 03:36 pm, Andrew Kohlsmith wrote:
exten = 1234,1,Dial(Zap/g1/5551234,,g)
exten = 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not
be BUSY?
Brian West pointed me
Andrei :
In zapata.confyou must activate the following lines
busydetect=yes
busycount=4
regards
Rodrigo
- Original Message -
From: Andrei (MPI) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, December 04, 2004
On December 4, 2004 12:59 am, Dinesh Nair wrote:
i've debugged the driver well enough and know that the Ouch message happens
when register 0x08 of the module returns 0, which indicates in most times
that digital loopback is enabled on the card. this register is set to
/disable/ digital
[EMAIL PROTECTED] wrote:
Hello,
I am using Mandrake 10.1.
Howto to install asterisk.
I have downloaded tarball.
I have not installed any hardware yet.
Is it possible to install ?
Yes,
http://www.voip-info.org/wiki-Asterisk
Doug
___
Asterisk-Users
snip
Use a good card like the 3ware 7500
series (parallel IDE ATA) and there
are no problems using IDE ATA drives. 3ware uses hardware raid unlike
the garbage promise chips that Claim hardware raid, but are not in
reality.
IED Raidsets on 3ware show up as scsi drives to the system.
3ware is one
Yes it is possible, you will running IAX and SIP phones only. You can get more
details about the installation www.voip-info.org
This are some good links to start.
http://www.voip-info.org/wiki-Asterisk+introduction
http://www.voip-info.org/wiki-Asterisk+installation+tips
Bye
Gerald
On Sat, 4 Dec 2004, Rich Adamson wrote:
The mind boggles -- PRI is *always* out of band.
Looks like the command is documented in the current config samples.
I'm not knowledgable/experienced (as yet) on where it is actually used,
but just reading the comments in the config sample led me
Ian Chilton wrote:
That's what I thought but I was told it was the only way to get incoming
SIP working when Asterisk was behind a firewall/NAT. I was told it was
not a security risk to do this.
If you *know* that only asterisk is listening on the relevant ports it's
less of a risk, but it's such
Carlos Clemares wrote:
Hi everyone,
I'm using the IAXy boxes and i'm having some trouble when I use it with
the ADPCM codec.
The IAXy only does ULAW
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Alan Ingleby wrote:
I also wanted to set up this machine to be our network
firewall/nat Our existing firewall runs linux on a p90, and runs
fine, but I figured it's time to upgrade.. Will this cause any
problems for *?
You might want to look into fli4l (http://www.fli4l.de). It
Same thing here. It used to work perfectly until I re-installed.
Hi
I need some advice in this issue, I installed astcc again and creates
database from configure menu but I am still getting errors messages:
in Brands menu: Something is wrong with the brands database
in Cards menu: Please
Hi, I cannot get SIP channel working with folowing codec configuration:
[sip]
disallow=all
allow=g723.1 ;I need this codec between sip phones (BT100)
allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge
(with g723.1) between them.
When
Hi,
TE Stack
No Upper ID
init_stack: File exists
You need to set the layermask when loading the card driver. For a TE
port, use 15 (layer 0-3) and for an NT port, use 3 (layer 0-1).
Simon
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[EMAIL PROTECTED]
Steve Totaro schrieb:
You might want to look into fli4l (http://www.fli4l.de). It is a
router/whatever plus there is a module add-on with asterisk. Might be
worth a try.
Is there a good site to check this out that is in English?
For fli4l itself, yes. For the opt_modul, no. After reading the
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
when it gets too busy :-)
I was wondering if it was poss to display this
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
when it gets too busy :-)
I was wondering if it was poss to display
test
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Jean-Louis curty to Asterisk
More options 4:38pm (7 minutes ago)
Hi everybody,
I'd like to know if anybody tried to write a xml doc to monitor the
number of calls in Q, when working with an ACD it's convenient to see
how many calls are waiting so the agent can speed up the conversation
I place a call from an IAXY to an IAXY device. INitially the calls show
in the output of show channels. Then after a few seconds the show
channels
command shows 0 active channels even though I am still talking on the
channels.
Any ideas on this?
THanks,
Jerry
I attempted this but I got stuck on one issue. Cisco phones pull data so I
couldn't get them to autoupdate. In other words push data to them. I am
working on an app to run on a windows desktop that will show the queues, the
amount of calls in each queue, the longest wait time and the average
My intention is to setup Asterisk to be a message center to receive
from and send SMS to fixed phones. Can it be possible? My fixed phone
can dial to Asterisk and send SMS to Asterisk, but I cannot setup the
other way: make Asterisk dial to fixed phone and send SMS to fixed
phone.
On Sat, 04 Dec
We have Grandstream SIP phones with the latest firmware versions and
have also have this problem. It appears to be something to do with RTP,
I believe. I don't know exactly what (simply because I don't know much
about RTP as yet), but the packets don't seem to reach the Grandstream
from the
Thomas Niesel wrote:
Hallo rich allen
I get this same error. Very strange. Dialing out from the IAXy works fine,
message:
Accepted AUTHENTICATED TBD call from 192.168.2.111
Accepted DIAL from 192.168.2.111, formats = 0x4
I also turned on Qualify and IAX debugging, and it reported my IAXy was
Sure, the IAXy's do a reinvite and * drops out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis
Sent: Saturday, December 04, 2004 10:50 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iaxy to iaxy call drops out of show channels
I place a
On Sat, 04 Dec 2004 10:49:42 -0500, Jerry Geis wrote:
I place a call from an IAXY to an IAXY device. INitially the calls show
in the output of show channels. Then after a few seconds the show
channels
command shows 0 active channels even though I am still talking on the
channels.
Any ideas on
Hi ALL;
I got the latest Asterisk-addons for Mysql-Cdr, but
I have problem compiling that.It says:
# make
.
res_config_mysql.c: In function
`realtime_mysql':res_config_mysql.c:143: warning: passing arg 1 of
`ast_strlen_zero' makes pointer from integer without a
Well, if these are the latest versio,ns of your files...
provision file:
codec: adpcm
iax.conf:
disallow=all
allow=ulaw
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asterisk wrote:
Hi, I cannot get SIP channel working with folowing codec configuration:
[sip]
disallow=all
allow=g723.1 ;I need this codec between sip phones (BT100)
allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge
(with g723.1)
-- Called 5899 at 192.168.0.5
-- Call accepted by 192.168.0.5 (format ULAW)
Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected
call to 192.168.0.5, format 0x4 incompatible with our capability
0xff03.
Hm, I'm not an expert on iaxY but it looks like that the
Leonardo J. Tramontina wrote:
No, I don't have anything connected on the TE110P.
After the Unable to create channel of type 'Zap' (cause 0) message,
I also get the CHANUNAVAIL...
Is not possible test a channel from the card without connections on it??
Leonardo
*snipped
no, the only way is if
i've debugged the driver well enough and know that the Ouch message happens
when register 0x08 of the module returns 0, which indicates in most times
that digital loopback is enabled on the card. this register is set to
/disable/ digital loopback upon an init.
the power alarm happens
Hi Walid,
Welcome to the list.
Zap are the connections from ordinary pstn
(or telco lines) to your digium hardware.
IAX is an Asterisk protocol for incoming
lines via IP from another asterisk PABX.
Hope this helps.
Dean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
B G wrote:
My intention is to setup Asterisk to be a message center to receive
from and send SMS to fixed phones. Can it be possible? My fixed phone
can dial to Asterisk and send SMS to Asterisk, but I cannot setup the
other way: make Asterisk dial to fixed phone and send SMS to fixed
phone.
Ah, I
Hi,
sorry for newbie Fritz question. I always thought that AVM Fritz has 2
devices for 2 MSNs. So does this mean, that Fritz can handle more ISDN lines
? Does this mean you can have more than 2 calls at once ? What is MAX
number of parallel calls ?
Thanks in advance,
Regards,
Robert.
-
Rich Adamson wrote:
Inline...
snip
Rich,
Thank you for your answer. Now I've figured that one of the FXO modules
on the card may be defective. Whenever I plug in telco line in it - that
line will be like shortened (if you pick up parallel telephone, the dial
tone will be heard weaker than
Typing trouble on my part. Should have said:
provision file:
codec: ulaw
iax.conf:
disallow=all
allow=ulaw
Elided email follows at end.
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent:
Hello,
I'm having an issue with native transfer not happening. I have a *
machine speaking ILBC in the middle of two * machines - everybody on
ILBC, but for some reason they will not transfer. All machines have
public IP addresses and can communicate directly with one another. One
thing I
Rich Adamson wrote:
The tdm card does have some unusual issues that appear to be driver
oriented, but there are lots of folks using the card in production.
Usually in situations where the client knows how to and tolerates having
to reload drivers and/or reboot his PBX periodically...
Regards,
Henry Devito wrote:
I attempted this but I got stuck on one issue. Cisco phones pull data so I
couldn't get them to autoupdate. In other words push data to them. I am
working on an app to run on a windows desktop that will show the queues, the
amount of calls in each queue, the longest wait time
Quoting Henry Devito [EMAIL PROTECTED]:
I attempted this but I got stuck on one issue. Cisco phones pull data so I
couldn't get them to autoupdate. In other words push data to them.
You can use an http Refresh to keep the screen updating once you've accessed
your XML application.
It's not
Hello!
I've encauntered some serious problems with asterisk.
I have to install it on system:
1. Mandrake 10.1
2. kernel 2.8.1
3. four ISDN cards.
And I am in big trouble,
isdn4linux is no longer supported for kernels 2.6 (on this system there are
not any /dev/ttyI0 and similar devices)/
Hi Rick,
If your configuration and firewall actually require you to open a
group of ports to *, then take a look at limiting the rtp ports that
are actually used.
How many do I need (or how do I find out?) and why does Asterisk specify
so many by default?
Thanks
--ian
No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones.
Message: 16
Date: Sat, 4 Dec 2004 12:45:53 +0200
From: "Walid Azab" [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and Cisco IP Phones
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type:
I, too would be very interested in this application. I have a small
call center with Cisco Phones, and one of our biggest problems is
alerting the Agents that a) there are calls in the queue; and b) they
have been logged out with the auto-logout feature. Most of our agents
converted from a
Corvin wrote:
Hello!
I've encauntered some serious problems with asterisk.
I have to install it on system:
1. Mandrake 10.1
2. kernel 2.8.1
3. four ISDN cards.
And I am in big trouble,
isdn4linux is no longer supported for kernels 2.6 (on this system there are
not any /dev/ttyI0 and similar
Hey All,
Quick Question. We just started using Remote-Party-ID
on our IAD endpoints and now when one of our customers has caller-ID blocked
(Privacy=full in the remote-party-id SIP header) and they call voicemail via
asterisks and get VoiceMailMain then they get a prompt for "Comedian
Pfft ya right if you want half assed supported channel drivers. Use SIP.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Keith O'Brien
Sent: Saturday, December 04, 2004 12:57 PM
To: [EMAIL PROTECTED]
Subject: RE:
Hi,
I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone.
However, I cannot find the SIP tab. Would someone please give me a few
pointers? The screen capture can be seen at URL below
http://www.dslreports.com/forum/remark,12022987~mode=flat
Regards,
Norman Zhang
Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND
SCCP unless you have actually installed and used it. Its crap...
SIP is what you want if you use a cisco phone with asterisk.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
On December 4, 2004 08:43 am, Rich Adamson wrote:
Looks like the command is documented in the current config samples.
Yeah I see that now. :-)
Since the comments use words like doesn't work with all telcos,
could this have something to do with detecting busy when a call
reaches a
Noah,
Thanks for the reply. I will try your instructions on Monday. I
appreciate it very much
Ferg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Friday, December 03, 2004 6:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
The * server is behind a Watchguard Firewall and I do have ports
forwarded. I will chyeck them on Monday. Thanks to all.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent: Saturday, December 04, 2004 10:54 AM
To: [EMAIL PROTECTED]
I do not have the Digium card on this box.
I have it on another box that I will eventually from it from.
All incoming calls are through IP and not any POTS line
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: Friday, December 03, 2004
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
Hello Ian,
VoiceMailMain(${CALLERIDNUM})
should do the trick (unless you have the blocked number problem a
previous poster had)
-yair
On Sat, 4 Dec 2004 20:01:58 +, Ian Chilton
[EMAIL PROTECTED] wrote:
Hi,
Is it possible to create an extension (say *1) that will give access to
the
Forgot the s
VoiceMailMain(s${CALLERIDNUM})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Saturday, December 04, 2004 2:08 PM
To: Ian Chilton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
true enough, forgot the s...the s skips the password
my bad
-yair
On Sat, 4 Dec 2004 14:14:06 -0600, Brian West [EMAIL PROTECTED] wrote:
Forgot the s
VoiceMailMain(s${CALLERIDNUM})
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Hi!
I want to install a X100P. I think I did everything according to the
manuals I found in the net. I loaded the modules and I edited the config
files according to http://www.digium.com/downloads/hw_article.
I start ztcfg that tells me everything is alright.
But when I start asterisk it tells
check it:
http://rpm.pbone.net/index.php3/stat/4/idpl/1516256/com/asterisk-chan_capi-0.
3.5-2mdk.i586.rpm.html
but I don't know if it resolve all problems .
Corvin
---
___
Asterisk-Users mailing
This is spam, but WOOT! I wonder if the bluez guys have put any
further work into fixing SCO.
On Sat, 4 Dec 2004 11:42:33 +, Mike Dent [EMAIL PROTECTED] wrote:
Thanks! :)
On Sat, 4 Dec 2004 10:19:59 +, Theo P. Zourzouvillys
[EMAIL PROTECTED] wrote:
On Saturday 04 December
--- Simon Richter [EMAIL PROTECTED] a écrit :
Hi,
TE Stack
No Upper ID
init_stack: File exists
You need to set the layermask when loading the card
driver. For a TE
port, use 15 (layer 0-3) and for an NT port, use 3
(layer 0-1).
Simon
Thanks, I
James Milne wrote:
Is there any workaround as of yet? Or is this something that polycom
will have to update in firmware?
It will have to be fixed in firmware, unless the problem is actually in
Asterisk; I do not know the actual cause of the problem. Unfortunately
since Polycom is not interested
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone
connection? I'd like to be able to connect using my cell phone data
connection, but so far I've come across bandwidth constraints - The closest
to success I've found so far is to use the GSM codec, but even then the
Hi,
I have Asterisk setup and registered with Gossiptel but i'm only getting
1 way audio.
If I call 160 (echo test) or 123 (talking clock), it makes the call but
I just get silence. If I call my Gossiptel number from a pstn line, I
get gossiptel - pstn audio but not pstn - gossiptel audio.
I've
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
Regards
Greg
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Phone - TDM430P - home* - IAX2 - office* - PRI - Telco
I dial a busy number from the Phone.
Home* shows this in the CLI:
-- Executing Macro(Zap/1-1, dial-wu|2922004) in new stack
-- Executing Dial(Zap/1-1, IAX2/[EMAIL PROTECTED]/2922004||g) in new
stack
-- Called [EMAIL
hi;
i have a conference room setup on the asterisk server. And say that one
of the sip peers (say A) wants to dial outbound to a PSTN destination
(say B). Can i have A join the conference room and some how at the same
time ask B to join the conference room?
I think this feature is very
Newbee here
I would like to play around with Asterisk a little.
First, I need to prepare a server with FreeBSD.
It's a PII 433mHz/256mb box. Good enough?
Then install Asterisk.
I have a broadband (cable) internet presence.
Could I do anything with this connection and
Asterisk?
Thanks,
If you get a Cisco phone, chance are it won't have SIP support right
off the bat, but you can upgrade the firmware to a SIP version. They
have the downloads available at their website but you have to buy some
sort of license/account with them. I've heard it's pretty cheap, but
I don't know first
Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've
followed the instructions posted by Broadvoice to configure sip.conf, and
inbound calling works fine. Every time I try to dial out, I get a 404 Not
Found error.
Here are the relevant sections from my configs.
Michael Vogel schrieb:
But when I start asterisk it tells me the error as posted in the subject.
The problem is solved. I had a version mismatch between zaptel and asterisk.
No I have different problems. But I will first try to find an answer in
the wiki before posting them.
Bye!
Michael
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