Hi,
Are there realy no-one who can help here
--
mvh. Hans-Henrik Andresen
--
Telefon for en flad 20'er - www.telefin.dk
--
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Hello
All,
So far I have been
unable to get the hard button labeled Voice Mail to conenct to Asterisk. I
have followed all the Admin Guide instructions regarding the .cfg files and
using up.bypassInstantMessage="1" up. to no avail.
Has anyone been able to get a Polycom 500 to use the
Wolfgang S. Rupprecht wrote:
Interesting. I'm at -current +/- a day and do see a
NAK/retry-with-md5 exchange when I do a sip debug. The md5
authentication is also NAK-ed.
Well you got farther than I got when I was having problems. :)
My fear was that it was expecting the calling user to use
It seems that way, I asked the same question about a month ago, and no one
cared to answer.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik
Andresen
Sent: 18 July 2004 07:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: MYSQL_FRIENDS
Hi All,
I have a asterisk box that is now on its own static address on the
net.it was originally behind a nat firewall.
The problem I have is that the remote SPA-2000's that are behind nat
firewalls now fail.
here is relevent sip.con entry
[2001]
type=friend
username=2001
host=dynamic
Hi,
On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new
Numbers. I changed the extensions in extension conf to match the new numbers. But i
always get:
Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel
'CAPI[contr1/89064934]/0' sent into
Hi All
Total noob on the list so all help appreciated
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm
looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones
The phones register with the Asterisk correctly
Hi,
-Original Message-
MSN messenger 4.7 with any windows capturing device should
work. Make
sure you force the codecs properly, because MSN tries to
negotiate some
form of MJPEG which Asterisk doesn't support.
How do you force the codecs? Do you do this in Messenger or
Hi,
-Original Message-
This is a little brief to say. I have had this working properly with
recent asterisk boxes. A few things: Check if the [general]
section has
'videosupport=yes' and if the sip peers are allowed to use h261 and
h263 codecs.
Best regards,
Florian
Hello There,
I tried checking out for this feature , what i want to do is that as
soon as the user picks up the handset , * waits for 10 secs and then
dials a predefined number , its like the HOTLINE feature we have in
normal POTs . Is it possible with Asterisk? If yes then how?
Regards
~uppal
On Sunday 18 July 2004 09:36, Junaid Uppal wrote:
I tried checking out for this feature , what i want to do is that as
soon as the user picks up the handset , * waits for 10 secs and then
dials a predefined number , its like the HOTLINE feature we have in
normal POTs . Is it possible with
it can also be defined on some devices like the grandstreams.
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 9:42 AM
Subject: Re: [Asterisk-Users] Hotline
On Sunday 18 July 2004 09:36, Junaid Uppal wrote:
I tried
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
As I couldn't get to compile and run Asterisk 1.0RC1 on my default
RedHat 9 I thought it was about time to upgrade to Fedora Core 2. Well,
it was too late to realize the kernel 2.6 wasn't supported by Asterisk
*officially* anyway.
Here is
On 17/07/2004 at 20:25 Josh Roberson wrote:
Seth Remington wrote:
I just updated from CVS and noticed that Mark has renamed all of the
parking related files (parking.conf, parking.h, res_parking.c) to
features.conf, features.h, res_features.c respectively. The CVS log
mentions that this is in
The reason is in the error message. Try using extension number 89064934
instead of 934.
Chris.
Tom Fischer wrote (on Jul 18):
Hi,
On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved
new Numbers. I changed the extensions in extension conf to match the new
I say on slashdot that the Linspire guys have released PhoneGaim.
PhoneGaim is Gaim with SIP added on. Anyone want to add IAX2 as
well...
http://www.phonegaim.com/faq.html
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I am trying to setup a simple pstn gateway using Asterisk and a X100p card.
I have got everything installed using Redhat 9 and am able to load Asterisk.
I also configured sip and I am able to connect to the asterisk gateway with
Xlite on the windows side.
I am able to dial 1000 and get the
I am running * with a Junghanns quadbri that should
allow us to integrate our ISDN house telephone system with VOIP. Preferably I
would like to run a setup, so that our internal ISDN phones on an S bus are not
aware that * is sitting in between.
With the configuration below I run into
Yes. I pulled the latest cvs and no cc in there at all anywhere. but
the bug http://bugs.digium.com/bug_view_page.php?bug_id=0001361 shows
that it is in fact committed to cvs.
I cannot seem to reopen this bug to say that it is not really committed.
I guess I will open a new bug report to do
try puttin this in extensions.conf
[outgoing]
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
and into your siphones extensions definition
[sip]
include = outgoing
Adrià Vidal
[EMAIL PROTECTED] | http://adria.homeip.net | MSN
[EMAIL PROTECTED]
iChat [EMAIL PROTECTED] | FWD [EMAIL
I added
exten = _0.,1,Dial,Zap/1/${EXTEN:1}
exten = _0.,2,Hangup
to the extensions.conf
but I am not sure I follow you on the second part, do you want me to add
include = outgoing
to my sip.conf file?? I did both of these changes, and I still have the same
problem.
Quoting Adria Vidal
There are many dyn dns clients for Windoze availible and some for linux
based computers. A few SOHO NAT routers support this also, but they are
limited in scope and may not work for your situation.
I think a workstation based solution is what you need if your router does
not support it.
Lyle
HI ALL;
I have couple of ip phones connected to my asterisk
box
1-cisco ata with sip protocol
2-sjphone with h323 protocol
as I understand, asterisk isable to translate
siph323 and vice versa ( am I right)???/
but when I try to
connectfrom ATA toSJPHONE and vice versa it
fails.
The readme says that the license uses all network cards MACS
What happens when VLANS are added or removed?
Is it safe?
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
I would comment out these lines in sip.conf
;externip=111.222.333.444
;localnet=192.168.1.0
;localmask=255.255.255.0
Then set nat=no
-Original Message-
From: Simon Chappell [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 4:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
On Jul 18, 2004, at 5:56 PM, Jason Armentrout wrote:
to the extensions.conf
but I am not sure I follow you on the second part, do you want me to
add
include = outgoing
to my sip.conf file?? I did both of these changes, and I still have
the same
problem.
must add
include = outgoing
into your
I just started out too and I can tell you it is easier to start from
scratch with a good wiki then alter the demo files. Here is a wiki you
can build a good working system with...
http://www.wlug.org.nz/AsteriskSampleSetup
For your ciscos search http://asterisk.xvoip.com/index.php
Wiley
On Sat, Jul 17, 2004 at 10:35:58AM +0200, Christian Ekhart wrote:
Hi,
we successfully use innovaphone IP200 H.323 hardware phones with OH323/Asterisk.
Calling/talking is OK, but call transfer does not work.
Does anyone of you use H323-phones with asterisk AND IS ABLE to perform CALL
It doesn't look like you have a context set for phone1. Try putting
context=sip in the phone1 section like you have in phone2. That'll put
both in the same context of your extensions.conf file and should allow
interaction between the two.
-Original Message-
From: [EMAIL PROTECTED]
hmm - this is the bad thing about open source etc.
Should we make a bugreport ? or are we just doing something wrong ?
--
mvh. Hans-Henrik Andresen
--
Telefon for en flad 20'er - www.telefin.dk
--
usedcanon
What I am NOT able to do is dial a seven digit local or 10
digit long distance number and make a phone call to the pstn
using the x100p card.
snip
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
What happens when VLANS are added or removed?
Is it safe?
Also, in this day of motherboard-integrated NICs (even two or three),
what will happen if the mobo dies and has to be replaced?
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
For a sollution to 1 ... drop the r option of dial...
exten = _X.,1,Dial(Zap/g1/${EXTEN})
You might need pridialplan/prilocaldialplan set to local for local
calls... or both to unknown...
hi all;
hi DANIEL;
I setup asterisk as a translator between sip-h323(I
used oh323 not native). But there is a problem and it is as
follows:
whenI try to dial FIRST from sip UA to h323
client, or h323 client to sip UA , it is ok
BUT the second try from any of
them to another have no
Hi Sean
Both phones are set for context=sip in the sip.conf file.
As I say the phones will both call out OK (I can dial the 500 test number and
successfully connect to the remote PBX through my firewall). It's just that when I'm
trying to call from phone to phone I'm getting the 404 not found
I can't think of any router that supports this
You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g
(great box...runs linux and lots of features and functionality).
P
-Original Message-
From: Lyle Giese [mailto:[EMAIL PROTECTED]
Sent: Sunday,
You can re-register the codecs one time using other NICS. after that
one time you need to contact Digium to be able to re-register, but the
process is very easy!
At 21:33 18.07.2004, you wrote:
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
What happens
I have a SIP phone that can make calls but can't recieve calls. Can anyone suggest why?
sip show peers:
Name/username Host Dyn Nat ACL Mask Port Status 601/601 (Unspecified) D N 255.255.255.255 0 UNKNOWN
-- Executing Dial("SIP/206.132.91.139-0814c9f8", "SIP/601|20|r") in new stackJul 18
Thanks for the tip, that made things work, it is really difficult for me to
understand the different config files and especially the extensions.conf, it is
very confusing. I am trying to learn though.
Now that I have got outgoing calls to work from the sip phone. How can I route
incoming calls on
Thanks for the tip, that made things work, it is really
difficult for me to understand the different config files and
especially the extensions.conf, it is very confusing. I am
trying to learn though.
Now that I have got outgoing calls to work from the sip
phone. How can I route
Your phone isn't registered. Ie Host
(Unspecified) so it has no idea where to send the call. Set your phone to
register and then asterisk can find it.
bkw
- Original Message -
From:
Joe
Babstock
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 2:57 PM
Thanks Marty,
That works now, the caller id on Xlite only shows the name for some reason, not
the number, but anyway it now rings in.
When I call the pstn number, the zaptel picks up the line on the first ring and
then forwards it to the sip phone and rings it. Is there anyway to prevent the
I had a Netgear WGR614 802.11g Wireless Router for a short time period,
it did support automatic dyndns updates, which was very handy.
Brian D'Arcy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 12:48 PM
To:
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
Our system is a just the basic subscription to SWISSCOM, which is the main
phone company in Switzerland. We have BRI with 2 Channels which can be used
simulaniously and a Siemens NT that has
Following a(n apparently) failed attempt to upgrade a BT102, the phone is
now brain-dead. Although it still has enough smarts to get a dhcp address
and try to download the firmware and config, it never gets past the blue
screen, nor will it respond to pings or port 80. Short of sending it back
Also, in this day of motherboard-integrated NICs (even two or three),
what will happen if the mobo dies and has to be replaced?
The same thing that would happen if the NIC died. IMHO it's a good thing
to tie to the NIC, because the chances of the MOBO dieing is not that
extreme. If it does
Marc Storck wrote:
You can re-register the codecs one time using other NICS. after that
one time you need to contact Digium to be able to re-register, but the
process is very easy!
That's good to know, thanks!
___
Asterisk-Users mailing list
[EMAIL
Brent Franks wrote:
The same thing that would happen if the NIC died. IMHO it's a good thing
to tie to the NIC, because the chances of the MOBO dieing is not that
extreme. If it does die, than just call digium and they'll re-license it.
Now it would be nice if when you install the codec, there
Bug report might be a good idea, I just dropped the issue as I could do
without using IAX. I am sure others may not have that flexibility.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik
Andresen
Sent: 18 July 2004 19:10
To: [EMAIL
Paul wrote:
Hi, i'm traying to compile asterisk on my pc, a laptop
whit SUSE 9.1 and a desktop with SUSE 8.2, with a teles S0
16/3 PnP. With Kernel 2.4 (Desktop) Asterisk run but
it's umpossible to compile the driver ISDN-utils for
Teles. With kernel 2.6 I can't compile zaptel (not necessary
with
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Sunday 18 July 2004 05:52 pm, Bruce Komito wrote:
Following a(n apparently) failed attempt to upgrade a BT102, the phone is
now brain-dead. Although it still has enough smarts to get a dhcp address
and try to download the firmware and config,
I am trying to compile chan_capi 3.3.4a, but I end up with lots of
gibberish. Near the top it states that capi20.h doesn't exist. Searching
for the file, several show up:
# find / -name capi20.h -print
/usr/src/linux-2.4.21-144/include/config/isdn/capi/capi20.h
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
The MAC address is unique a 6 byte address assigned to every 802-family
(802.1 Ethernet, 802.11 wireless, etc.) network interface.
What happens when VLANS are added or removed?
Nothing... VLANs have absolutely no
Hallo, due to everchanging CVS,
chan_capi-0.3.4a doesn't compile anymore with new cvs
my solution was to chande chan_capi.c
the line 21 from
#include asterisk/parking.h
to
#include asterisk/features.h
now chan_capi compiles again and seems back on duty again.
Hope this help.
Diego
I have an Avaya 4602SW SIP phone.
They just released the SIP firmware for it the other day.
I have it working with my Asterisk, but have a couple issues.
My setup is like this: Avaya 4602 phone at home behind router and Asterisk server is
straight on the Internet.
My phone registers with
On 03:33 PM 7/18/2004, usedcanon wrote:
Bug report might be a good idea, I just dropped the issue as I could do
without using IAX. I am sure others may not have that flexibility.
Umar.
-Original Message-
Subject: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem
hmm - this is the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
On 19/07/2004, at 9:08 AM, Thor Atle Rustad wrote:
I am trying to compile chan_capi 3.3.4a, but I end up with lots of
gibberish. Near the top it states that capi20.h doesn't exist.
Searching for the file, several show up:
Make sure that you've
Hello
All,
I have some Polycom IP 500 phones that I would like to
have configured for direct dialing to our voice mail system.
So far I have been unable to get
the hard button labeled Voice Mail to connect to Asterisk
without first passing through the message center prompts. I have
Hello
I am trying to setup ChanIsAvail function in the
extensions.conf file so that user should use the available channel to call out,
but immediately after the function like, zap channel hangup.
Here is the copy of my extensions.conf file and
messages display on consol while making the
I believe that 'ast_data' is the solution to this problem, and will
probably obsolete mysql friends. However, I could be incorrect in that
manner. There are folks on this list who would be much better informed to
say whether or not it will obsolete mysql friends.
-Chris
I did not tests
Nicholas Bachmann wrote:
Anton Tinchev wrote:
The readme says that the license uses all network cards MACS
The MAC address is unique a 6 byte address assigned to every 802-family
(802.1 Ethernet, 802.11 wireless, etc.) network interface.
What happens when VLANS are added or removed?
Nothing...
Wiley E. Siler wrote:
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center
So are saying that T2240 will gurantee no echo issues? Did you get any
echo issues with a different PC with the same cards and Pstn lines?
snip
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
I'm thinking of doing an app to work with the CID that's gotten from
the Zap channel.
All the CID's I see from within the US are 10 digit numbers.
I'm out in the rural areas of the US, and no-one ever calls me from
overseas.
If they did, what would the CID look like?
What does the CallerID
Hello,
I haven't seen any recent posts on call progress detection, so here's a question:
How would one accomplish an automated outbound dialing application using *, whereby a
requirement is to wait for the greeting to complete (live person, answering machine,
voicemail) before delivering the
On Sun, 2004-07-18 at 20:38, Stephen David wrote:
Hello,
I haven't seen any recent posts on call progress detection, so here's
a question:
How would one accomplish an automated outbound dialing application
using *, whereby a requirement is to wait for the greeting to complete
(live
Does anyone have a recommendation for a 48 port LAN switch for a new *
system? I'm not happy with NetGear's reliability.
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado
+1 303 674 2575
[EMAIL PROTECTED]
www.introspect.com
___
Make sure that you've created a link from /usr/src/linux-2.4.21 to
/usr/src/linux
ln -s /usr/src/linux-2.4.21 /usr/src/linux
then recompile asterisk
The symlinks were already there.
# ls -ld /usr/src/linux*
lrwxrwxrwx1 root root 25 Jul 19 03:46 /usr/src/linux -
On Sun, 18 Jul 2004, Michael Welter wrote:
Does anyone have a recommendation for a 48 port LAN switch for a new *
system? I'm not happy with NetGear's reliability.
You can get Cisco 2950s for about $600/24 ports.
___
Asterisk-Users mailing list
On Jul 18, 2004, at 7:14 PM, [EMAIL PROTECTED] wrote:
On Sun, 18 Jul 2004, Michael Welter wrote:
Does anyone have a recommendation for a 48 port LAN switch for a new *
system? I'm not happy with NetGear's reliability.
You can get Cisco 2950s for about $600/24 ports.
And 48 ports from Dell for
I'm installing TE405P card.
This is my zaptel.conf.
--
span=1,0,0,ccs,hdb3,crc4span=2,1,0,ccs,hdb3,crc4span=3,0,0,ccs,hdb3,crc4span=4,0,0,ccs,hdb3,crc4
loadzone = usdefaultzone=
us--
When i modprobe
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've forgotten the command to add a vm box, and searching google and wiki I'm
surpriced I cannot find it. I'd love to know where this is written, so I can
see how I managed to miss it!
- --
Steve
They that would give up essential liberty
I have been quite happy with our HP 2848 GigE switches that we put in
for our desktops a few months ago. I have also used the 2650 48 10/100
+ 2 GigE switches before.
We are looking at the 2650-PWR for our VoIP deployment (only about 60
phones for our USGS/U of A mixed department).
Hi,
I've forgotten the command to add a vm box, and searching google and wiki
I'm
surpriced I cannot find it. I'd love to know where this is written, so I
can
see how I managed to miss it!
- --
Steve
Look for your controb/script directory. The script is called 'addmailbox'.
Regards,
On Sunday 18 July 2004 11:21 pm, CW_ASN wrote:
Hi,
I've forgotten the command to add a vm box, and searching google and wiki
I'm
surpriced I cannot find it. I'd love to know where this is written, so I
can
see how I managed to miss it!
- --
Steve
Look for your
When I call the pstn number, the zaptel picks up the line on
the first ring and then forwards it to the sip phone and
rings it. Is there anyway to prevent the zaptel from picking
up the line until the sip phone actully answers the call.
This way I could answer the phone either locally on a
Thanks for that.
Like many I believe * is unusable in production until these echo issues are quoshed are resolved. Lets hope someone takes up the bounty offer.Rich Adamson [EMAIL PROTECTED] wrote:
So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different
Hi all,
I have the following setup:
UAs SER -- ASTERISK --GNUGK - GWs
SER is configured to route call requests from UAs to Asterisk. Asterisk is
configured to receive the call on SIP channel and dial out to GNUGK over
H323 channel. The problem I'm facing is
Hi,
I am Abhishek from India.
I am have studying Cisco VOIP since a couple of months.Searching for Soft
PBX somenthing like (Cisco Callmanager) i came accros this Asterisk.
I have to provide a a solution to a clinet where he wants a connectivity
between his 3 offices across the WAN with a very
Abhishek,
In reverse order
3/ yes it is freeware, though some of the termination boards are
available for sale from www.digium.com
2/ yes you can interface to Cisco handsets running SIP.
1/ Does it have a gui interface - the short answer is no.
The longer answer is depending on what you mean,
I have a solution that allows me to assign a soft key with no problems.
However, it seems like a waste the the hard button labeled Voice Mail is
not dialing right into voice mail. Is there a known way yo do this? I
have tried everything in the manual but it doesn't seem to work. I have
IP 500s
Does anyone know
where I can find a list of all the control scripts? I want to write a
standard windows tool that will allow you to pregenerate the configuration for
your Asterisk install and them press one button to have it log into your
boxand upload the scripts. Of course, I will let
Dont have to.. just add it to the voicemail.conf and it will auto do
everything for you.
bkw
- Original Message -
From: Steve [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 9:58 PM
Subject: [Asterisk-Users] Adding voice mail box
-BEGIN PGP SIGNED
For those who don't watch asterisk-cvs, it appears that markster has
begun (and possibly) completed adding GR-303 FXS support to Asterisk.
This means that Asterisk could be used as an access concentrator off
of a class 5 switch, which gives us a higher-level alternative between
using single
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