Hi Olle,
that was a phone misconfigurationa parameter had a wrong value.
The message has disappeared and now the phone seems to work!
Thank you!
Giorgio
Johansson Olle E wrote:
10 jan 2008 kl. 16.48 skrev gincantalupo:
Hi,
I'm using an Asterisk 1.2.18 box with a remote Snom 360. My
there is no /proc/zap folder .. can you tell how can I create /dev
nodes. I have tested the same configurations on FC5 and these device
links were created ...
drwxr-xr-x 2 root root 160 Jan 17 10:59 .
drwxr-xr-x 13 root root 3640 Jan 17 11:00 ..
crw--- 1 root root 196, 1 Jan 17
Can anyone share their experience with me? I am looking for a provider that
delivers Dialtone over T1 to terminate to my asterisk box and also provide
DIDs. Does the DIDs come with the T1 services or those are purchased/charged
seperately. Any help greatly appreciated. My target markets are
*Walter Willis,
*Thanks a lot, got the commands from zap Makefile and it worked, now can
create conference room, my question still stands why it didn't create
itself. Will go through make file to get an answer to that.
Anyone else facing the issue can resolve by running following commands
Hello.
I have a little problem with the callerid shown to the callee if he recieves
an atxfer (*2) call. The display of the calees phone is showing (s) and thats
not what i want. I wanna see the callerid from the user who is transfering
the call. Example:
12345 calls 123, 123 transfers
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
However, you'll need to do similar things to your asterisk
box router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and
On zapata.conf use the parameter callerid.
On Jan 17, 2008 3:33 AM, sandeep [EMAIL PROTECTED] wrote:
hi all,
how to set the caller id facility for
the TDM400p card.
Please help me
thanks,
sandeep.s
--
Guilherme Loch Góes
Visite nossa loja virtual: http://www.shopvoip.com.br
Notícias
Hi everyone,
I have been long working on a project (http://asterisktools.org, to be
released under GPL) that aims to provide desktop tools for Macs. I am
finally getting to the release stages of this application and hope to
have an early BETA available next weekend.
If there is anybody who is
On Wed, Jan 16, 2008 at 10:09:54PM -0500, Walter Willis wrote:
any version of asterisk not create nodes into /proc/zap
create to command, view into make file how to create nodes
Do you suggest to use mknod manually?
This will work. Unless you use udev. And almost everybody use it.
What is the
Hi All
I need to set my Asterisk conference such way that , during
confernce Admin Can kick 1 or all user , Same for mute fuction.As well as
Admin can increase or decrease conf user volume.
for that i used MeetMeAdmin like this
exten
Hi,
Im interested, Please send me copy
Thanks
On Jan 17, 2008 7:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Hi everyone,
I have been long working on a project (http://asterisktools.org, to be
released under GPL) that aims to provide desktop tools for Macs. I am
finally getting to the
Looks interesting. I couldn't get it working because a few of the
preference fields were not responding (current svn, build on Leopard).
Looks like a nice elegant solution though. Let me know if there's
anything you want help on and I'll dust off my cocoa!
Simon
Simon Elliston Ball
[EMAIL
On Thu, 17 Jan 2008 6:34 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
And now in make menuselect you have to go to voicemail options and set IMAP
support to on.
Thanks, if that was in any of the docs I just completely glossed over
it. I'll give it
a shot.
Thanks again,
On 1/17/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote:
Hi,
I'm wondering why zttest shows
Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469
Shouldn't it be 100% as timing is hardware and comes from PRI?
Hi,
I'm wondering why zttest shows
Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469
Shouldn't it be 100% as timing is hardware and comes from PRI? Am I
missing some kernel config?
Regards,
Atis
My /etc/zaptel.conf is
span=1,4,0,esf,b8zs
span=2,3,0,esf,b8zs
On Jan 17, 2008 5:23 AM, broadband Voice [EMAIL PROTECTED] wrote:
Can anyone share their experience with me? I am looking for a provider
that delivers Dialtone over T1 to terminate to my asterisk box and also
provide DIDs. Does the DIDs come with the T1 services or those are
purchased/charged
voip*CLI ael reload
Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
root token '#include'
Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box,
and I don't want to upgrade our only production computer.
Jay
Rodrigo R Passos wrote:
Jay,
What error?
On 1/17/08, Jay Moore [EMAIL PROTECTED] wrote:
How do I include a file (not a context) in AEL? #include filename
returns an error.
What's the error?
For me this works:
#include extensions_db.ael;
#include extensions_utils.ael;
#include extensions_ivr.ael;
#include extensions_globals.ael;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jay Moore
Sent: Thursday, January 17, 2008 9:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AEL includes?
How do I include a file (not a context) in
How do I include a file (not a context) in AEL? #include filename
returns an error.
Thanks,
Jay
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On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote:
Hi,
I'm wondering why zttest shows
Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469
Shouldn't it be 100% as timing is hardware and comes from PRI? Am I
missing some kernel config?
It may be slightly
Steve,
That is very helpful, How much are we talking about in terms of the loop and
minute charges. If you want it offline I can send you a private my with my
phone number.
On 1/17/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Jan 17, 2008 5:23 AM, broadband Voice [EMAIL PROTECTED] wrote:
Hi,
When asterisk receives 302 Moved Temporary sip response what is the logic
for selecting the domain and context to use?
Thanks for any help
Tomasz
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Jay,
What error?
Jay Moore wrote:
How do I include a file (not a context) in AEL? #include filename
returns an error.
Thanks,
Jay
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To
On 1/17/08, Jay Moore [EMAIL PROTECTED] wrote:
voip*CLI ael reload
Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
root token '#include'
Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box,
and I don't want to upgrade our only production computer.
Cavalera Claudio Luigi wrote:
Is this the libiax used currently on asterisk
http://ftp.digium.com/pub/libiax/ ?
No. Asterisk has its own IAX2 implementation.
I would like to understand if someone is using this in production.
I have no idea if anyone is using it. It's easy to use, so I
This is just a low priority curiosity question because I have a usable
workaround.
I have Digium card that uses the Zaptel driver (can't get to my home
machine right now to get the exact model, but it probably doesn't
matter). It's a card with one POTS line and three extension hookups. I'm
using
The public Asterisk SVN mirror is back up to date. I apologize for the
inconvenient downtime. Re-syncing with a repository that has almost 100,000
revisions took a while. :)
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
Hello,
from what I've understood Iax2 should support aes128 encryption.
I've found this old info:
http://www.voip-info.org/wiki/view/IAX+encryption
and this (unanswered?) post
http://lists.digium.com/pipermail/asterisk-security/2005-August/60.h
tml
Is this the libiax used currently on asterisk
AEL was an experimental feature in Asterisk 1.2.x and you may not implement all
funcionts.
Jay Moore wrote:
voip*CLI ael reload
Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
root token '#include'
Asterisk 1.2.14. Old, I know but my boss won't spring for a spare
Hi,
I'm wondering - why SIP device state doesn't get updated to anything
else, except Not In Use.
For queue call (with Local channel) i get:
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
app_queue.c: The device
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Russell Bryant
I would like to understand if someone is using this in production.
I have no idea if anyone is using it. It's easy to use, so I
assume that some
people are ...
I guess what
Cavalera Claudio Luigi wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
However, you'll need to do similar things to your asterisk
box router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
Thank you all for the voicemail cards you sent.
If you have the following in PDF or laying around (scan):
* ATT/Cingular flow voicemail card
* Verizon flow voicemail card
* Sprint flow voicemail card
* TMobile flow voicemail card
* Alltel flow voicemail card
* Avaya Nortel Octel flow voicemail
Greg Woods wrote:
This is just a low priority curiosity question because I have a usable
workaround.
I have Digium card that uses the Zaptel driver (can't get to my home
machine right now to get the exact model, but it probably doesn't
matter). It's a card with one POTS line and three
TMOB
http://support.t-mobile.com/knowbase/root/public/tm22131.htm
Thanks,
Steve Totaro
On Jan 17, 2008 1:54 PM, Justin Newman [EMAIL PROTECTED] wrote:
Thank you all for the voicemail cards you sent.
If you have the following in PDF or laying around (scan):
* ATT/Cingular flow voicemail
Yaah!!! Mac! I am a big user of OS X. Can't help it. Macs eye candy draws
me in like my wofe. :) And.. I've never had a single issue with it. I also
host virtual Ubuntu, Red Hat and XP :( on the same box using VMware.
Sorry about the Mac rant. Just glad to see some Mac / Asterisk
Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls
working fine, the key is making sure the channels are bridging and EC is NOT
turning on. If you have anything preventing that the modem calls won't work.
-Original Message-
From: [EMAIL PROTECTED]
On Jan 17, 2008 2:28 PM, Nicholas Blasgen [EMAIL PROTECTED] wrote:
I've set up plenty of Asterisk boxes but never one that had to deal with a
proxy server to be able to use a line. Using X-Lite I have no issue with
settings as follows:
Display Name: Any Name
User name: 0057510
I am querying an postgresql database from my 1.4.13 system and the results
seem to be truncating each column at 255 characters. The columns are typed
as character varying 1000.
Any suggestion on how to remove this limit?
TIA
Vic
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Hi all
Someone has make a voicemail callback on * ?
Thanks
--
Gilberto Nunes
Itajaí - SC
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I've set up plenty of Asterisk boxes but never one that had to deal with a
proxy server to be able to use a line. Using X-Lite I have no issue with
settings as follows:
Display Name: Any Name
User name: 0057510
Password: 0057510
Authorization user name: blank
Domain:
On Jan 17, 2008 1:28 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem
calls working fine, the key is making sure the channels are bridging and EC
is NOT turning on. If you have anything preventing that the modem calls
won't work.
Hi Folks,
I'm currently trying to configure musiconhold (on a asterisk-1.4.17) for
replaying a live mp3-stream (Icecast2). after reading the related material
on voip-info and several other pages, I've successfully tried out mpg132,
madplay and mplayer to pipe a stream into moh.
however, there is
Atis Lezdins wrote:
Hi,
I'm wondering - why SIP device state doesn't get updated to anything
else, except Not In Use.
For queue call (with Local channel) i get:
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in
I'm interested too Devraj, please send a copy of if possible to try it.
Thanks.
On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Hi everyone,
I have been long working on a project (http://asterisktools.org, to be
released under GPL) that aims to provide desktop tools for
What are people's thoughts on asterisk 1.2.26? Any show stopping bugs?
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Thanks for your response guys. There are still some issues with the
code (Svn on SourceForge). I am working on getting these fixed up and
will post a message when its ready for download.
I will yell out if I need some Asterisk/Cocoa help. Thanks a lot.
On Jan 18, 2008 7:19 AM, Adrià Vidal [EMAIL
thx a lot russel...your hack actually works!! :)
Meanwhile I've found something about the musiconhold-conf-option
cachertclasses, which might help in starting a separate instance for every
caller. however, that didn't really work for me... probably this option only
works for mode=files?!
Hi Tzafrir,
Yes it does use the Manager Interface. It account does require call
level access. That may then result in umlimited access to Asterisk
(well to originate calls anyway). However I have made real conscious
efforts to filter messages that are being transmitted over the socket
so the
I am connected to CCM and have a sip.conf entry like:
[CCMHEART]
type=friend
host=X.y.X.A
allow=ulaw
allow=alaw
allow=all
canreinvite=yes
qualify=yes
context=CCMHEART
In extensions.conf I have a context of:
[CCMHEART]
exten = s,1,Goto(default,s,1)
exten = 45801,1,Goto(default,s,1)
exten =
I am looking to see if anyone has seen this problem before. I am
setting the MEETME_RECORDINGFILE variable in a macro, then using the r
option with the Page application to record the page. But the page is
only recorded to the file specified in MEETME_RECORDINGFILE
sometimes...
This is my configuration in the extensions.conf,
iax.conf at Site A and Site B, so anyone can help why
the call refused?
Site A:
[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.3
usename=IPLink
secret=password
canreinvite=no
nat=no
[SiteBInternal]
exten = _2XX,1,Dial(IAX2/[EMAIL
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
by accident and is away from the phone. Does anyone know of a way to
remotely un-forward the phone? I tried to reboot the phone but that
didn’t work and removing the mac-phone.cfg caused problems
I have a remote user on a Polycom IP Phone who has set call forwarding by
accident and is away from the phone. Does anyone know of a way to remotely
un-forward the phone? I tried to reboot the phone but that didn't work and
removing the mac-phone.cfg caused problems
Michael Kamleitner wrote:
10:00 I'm calling the pbx, musiconhold starts correctly to play the
live-stream (almost live, with very small delay) - that's OK.
10:01 I hangup.
-- than I pause for 20 min --
10:20 I'm calling a second time. However moh now doesn't stream live, but
starts to
On Jan 17, 2008 7:55 AM, KodaK [EMAIL PROTECTED] wrote:
Thanks, if that was in any of the docs I just completely glossed over
it. I'll give it
a shot.
Yes, I skipped over that in the docs. I'm good at that.
Thanks for the help.
I've also written up a quickie how-to on how to enable this
Michael Kamleitner wrote:
thx a lot russel...your hack actually works!! :)
Awesome. :)
Meanwhile I've found something about the musiconhold-conf-option
cachertclasses, which might help in starting a separate instance for every
caller. however, that didn't really work for me... probably this
On Wed, 2008-01-16 at 15:52 -0800, Steven wrote:
I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
TE220B. The system load is below 0.10.
I moved the server into production, with one PRI, on Friday.
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I
re-touched the file and followed your suggestion.
Any way of removing the call forwarding feature via the xml configs?
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
by
I have a suggestion. Have you contacted Digium technical support
for assistance
with resolving this issue?
Excellent suggestion. Make sure you can give them SSH access and
screen so you can see what they are doing. Before that, check
(remake) your T1 cables and if it is
Andrew Joakimsen wrote:
I'll assume you chose trixbox to make your life easier when it comes to
dealing with others
regarding the PBX.
Pretty much, yes.
What is between the smartjack and your T1 card? What sort and length
of cable? Any splices? Punchdown or patch panels?
About 100
When setting a forward on the phone, the phone will upload to your ftp
server a modified macaddr-phone.cfg XML file that (amongst other
locally made changes) contains an OVERRIDE statement similar to this:
OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... /
Change the .fwdStatus attribute to
Hi there
this is an interesting topic that I see here and a problem that I am trying to
solve too.
But I was wondering if the forwarding solution will work for my case.
So I have two Asterisk boxes A and B.
A is behind a corporate NAT such that A can SSH to B, but not vice versa(
You mention went into production, Did this imply moving of the system
from a testing room into a server-location? Other (longer) cables?
Unplugged the current system and hooked up a new, longer, cable to the
asterisk system. The cable is RJ48 STP, about 100 feet. However we ran
several
Dear all.
I have about 30 Cisco 7910 handsets, and my basic research has told me that
they are not SIP based handsets. Not to worry for now, I just need them to
connect to my asterisk server. They are giving me a bit of a hard time. Has
anyone here had any experience on how to do this?
Dear all,
I have managed to connect this device to my asterisk box, but it is giving
me a bit of a hard time. I can call other extensions from this box, but I am
not able to call this one. It seems to permanently remain engaged. When I
dial it, this is the message I get. Is there a know issue
On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote:
Hence, is my only choice using an SSH tunnel between A and B for the
IAX connection to work? Will it work though with that One-way SSH
factor mentioned before?
It's my understanding that SSH tunneling will only work with TCP
traffic.
On 1/18/08, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote:
Dear all,
I have managed to connect this device to my asterisk box, but it is giving
me a bit of a hard time. I can call other extensions from this box, but I am
not able to call this one. It seems to permanently remain engaged.
Diax is probably the smallest Windows softphone.
Add to that list Mozphone (http://mozphone.mozdev.org/) that can be
installed in Firefox
Kiax : http://sourceforge.net/projects/kiax
shameless plugMy MediaX softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless
plug
iaxcomm
I guess I was interested in Disabling the forwarding feature completely via
the config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Thursday, January 17, 2008 7:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I misread then. Even though your original message said you wanted to
un-forward a phone. That can be done with the recipe BJ and I
outlined.
I am not aware of any way to disable the forward function, i.e.
prevent a user from forwarding in the first place.
Jared Smith wrote:
On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote:
Hence, is my only choice using an SSH tunnel between A and B for the
IAX connection to work? Will it work though with that One-way SSH
factor mentioned before?
It's my understanding that SSH tunneling will only
In your per-phone configuration:
phone1
reg
...
divert
divert.fwd.1.enabled = 0
divert.fwd.2.enabled = 0
divert.fwd.3.enabled = 0
divert.fwd.4.enabled = 0
divert.fwd.5.enabled = 0
divert.fwd.6.enabled = 0
/
This removes the soft-key and
Good question. I have never tried tunneling IAX over SSH but it seems like
it should work just like anything else.
How about a port opened up for OpenVPN. You know you can run IAX on any
port you wish, port 80 may work for you if you have some extra external IPs
not being used for HTTP. The
What message? NAT?
On Jan 17, 2008 8:18 PM, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote:
Dear all,
I have managed to connect this device to my asterisk box, but it is giving
me a bit of a hard time. I can call other extensions from this box, but I am
not able to call this one. It seems
There is no NAT involved, just a straight connection
Mr Gabriel Ogunleye
IT Administrator
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: 18 January 2008 05:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
There is no NAT involved. I think I will try to sip set debug. What
exactly should I be looking for?
How did you configure these devices - maybe something I missed in the
config?
Mr Gabriel Ogunleye
IT Administrator
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
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