In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
It looked promising so I tried it. Unfortunately it didn't help. Calling
person doesn't hear ringing. I don't know why this application didn't work as
it should. I have tried with and without wait command.
-- Executing
On 09/19/06 16:59 Steve Langstaff said the following:
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124nbn=4
thanks for the link,
however, on 18th may 2006, kpfleming's note says, This should be
gateway sip mysipprovider
no transport tcp
bind interface WAN router
domain mysipdomain
realm sip.mydomain.nl
authentication myusername password mypassword
default-server mysipproviderserver 5060 loose-router
registration-lifetime 300
registrar mysipproviderserver
Hi all,
trixbox has taken control of my asterisk system, i dont like that. i just installed trixbox for rersearch purpose now i want to uninstall it and do some research on asterisk. So plz tell me how to uninstall trixbox. will it uninstall asterisk also?
-- RegardsRizwan HishamSoftware Engineer
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi all,
I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes
starts eating up a lot of CPU.
I sthere any alternative method to use moh without use mpg123?
I tryied this http://astrecipes.net/?n=152 but i doesn't wotks for
>From my understanding, tribox is to control asterisk via web interface. so, if u want to uninstall the tribox, i guess just delete the web folder will do then do can edit direction frm your asterisk files.
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
trixbox has taken control of my
The following works for me, for e.g. extension number '5301' with a
secret of 'secret' on server '192.168.1.1'
Addressing
SIP Address-of-Record (AOR): sip:[EMAIL PROTECTED]
Registrar Server
Domain: asterisk
Expiration: 3600
Authorisation
Update Authorisation: checked
Username: 5301
Realm:
No, it says I don't know what to do if the via is *not* SIP/2.0/UDP
Yes, my error.
I tested with asterisk 1.2.10 and now it works.
Reason: Steve Langstaff found that in my old * version there was case
sensitive strcmp which caused the problem.
Thanks to you all,
Christian
Subject: RE:
On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote:
On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote:
michael,
at my real job, the phones display peoples names when calling out from
your phone. how is this done?
Maybe they put the names in the phones internal addressbook
?
--
On Wed, Sep 20, 2006 at 01:07:11PM +0500, Rizwan Hisham wrote:
Hi all,
trixbox has taken control of my asterisk system, i dont like that. i just
installed trixbox for rersearch purpose now i want to uninstall it and do
some research on asterisk. So plz tell me how to uninstall trixbox. will it
hi
is there any software usable to simulate work on an asterisk server?
I'm interested in it to evaluate the level of currently calls that a
server can support
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To
On Tue, Sep 19, 2006 at 06:53:01PM +0200, Giordano Grandis wrote:
Hi all,
I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes
starts eating up a lot of CPU.
I sthere any alternative method to use moh without use mpg123?
I tryied this http://astrecipes.net/?n=152 but i doesn't
Hi All
I'm tracing a strange BRI Q.931 Problem with Asterisk 1.2.4.
I call a number which is diverted to another number.
Asterisk seams to take this divertification message as a hangup message:
BRI Trace:
-- Executing Dial(IAX2/magma-1, Zap/g7/0418103734|90) in new stack
2 -- Making new
Le mardi 19 septembre 2006 à 15:30 -0500, David R. a écrit :
Can AGI be used to have a web application talk back and forth between
Asterisk and itself? What if the web application is on a separate
box?
As Stefan Reuter previously stated, there's no problem running your AGI
application
no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE.
By saying that 'It has taken control of my system', i meant asterisk.
Now i dont want any web based interface to asterisk. i only want
asterisk on my system.
hehehe, then you have to edit the startup manual. I dont think so there is a way to uninstall the tribox. for example to disable start up for asterisk, i think you can try type the command chkconfig asterisk stop ..good luck
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
no, im not using
Hi,
I have a problem with Asterisk AGI command.
I wrote a script which launches a shell command.
If I launch a normal command for example like ll /tmp/tmp.txt, the
AGI command launches the shell commands and then exits.
The problem is when I launch THIS command to create an ssh tunnel in
hi asterisk' ians How to write agi scripts, how to see the output.. thanks in advance...
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
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The radvision's prolabs is your best choice for SIP or H.323.
2006/9/20, nik600 [EMAIL PROTECTED]:
hi
is there any software usable to simulate work on an asterisk server?
I'm interested in it to evaluate the level of currently calls that a
server can support
On 9/20/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I have a problem with Asterisk AGI command.
I wrote a script which launches a shell command.
If I launch a normal command for example like ll /tmp/tmp.txt, the
AGI command launches the shell commands and then exits.
The problem is
On Wed, Sep 20, 2006 at 02:53:04PM +0500, Rizwan Hisham wrote:
no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc. and i also dont know about CentOS, im using RHL EE. By
saying that 'It has taken control of my system', i meant asterisk. Now i
dont want any
Hi,
I am looking for some docs to help configure a AudioCodes Mediant 1000
with asterisk, any tips or examples are appreciated.
raj
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Thanks Sharon.
On 9/20/06, Sharon Lim [EMAIL PROTECTED] wrote:
I have successful link skype with asterisk with
http://www.nch.com.au/skypetosip/index.html but not sure
whether you need this.
here is another link
http://www.voip-info.org/wiki/index.php?page=Skype%20Gateways.
Good luck!
On
good stuff mate.
a few clarifications:
you had static extensions.conf, realtime sipusers, etc, right?
Also, abt features like call fwding, etc, which one is better,
performance wise, using a mysql db, or use Asterisk's internal
DB(berkeley db, isnt it?using those DBput n DBget
I would like to know how you got Asterisk to function with 2500 SIP
registrations. Did you have qualify enabled?
Yes, qualify was enabled, using the standard length of qualification
period between checks. Very few accounts had custom qualify settings.
What about the 500 simultaneous
When you installed Trixbox did you not boot from the Trixbox install CD?
This installs CentOS and Trixbox.
I'm curious how you installed Trixbox?
Mike
On 9/20/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
no, im not using FreePBX, actually freepbx is a part of trixbox as is
sugarCRM, FOP etc.
Sir,
I installed asterix in some system say with ip 172.16.7.63.From some other windows system say 172.16.7.50 i am running an xlite and configured a user say 200 with proxy as 172.16.7.63.I
modified the sip.conf file with user [200].When i run xlite in asterix cli i am able to see the mesage
On 9/20/06, nik600 [EMAIL PROTECTED] wrote:
hi
is there any software usable to simulate work on an asterisk server?
I'm interested in it to evaluate the level of currently calls that a
server can support
For SIP, see http://sipp.sourceforge.net/
--
Bird's The Word Technologies, Inc.
Sir, I installed asterix in some system say with ip 172.16.7.63.From someother windows system say
172.16.7.50 i am running an xlite and configured auser say 200 with proxy as 172.16.7.63.I modified the sip.conf file withuser [200].When i run xlite in asterix cli i am able to see the
I Installed it using the rpm that is available for download on trixbox website along with the ISO image. Well actually its on sourceforge.net. anyways, unlike the ISO image, rpm package only installs trixbox and asterisk not the operating system. its for linux.
On 9/20/06, Mike Dent [EMAIL
Mike Dent wrote:
I'm curious how you installed Trixbox?
There is a tar.gz version of Trixbox that can be installed over an
existing RHEL4 or CentOS installation.
However, removing Trixbox is very difficult. You are better off
reinstalling RHEL4 and then installating Asterisk from scratch.
rpm -e packagename ?
Rizwan Hisham wrote:
I Installed it using the rpm that is available for download on trixbox
website along with the ISO image. Well actually its on sourceforge.net
http://sourceforge.net. anyways, unlike the ISO image, rpm package
only installs trixbox and asterisk not
G'Day
List,
Interesting article.
Enjoy
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5
Mike
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sorry i made a mistake telling you that i installed it using rpm package. actually there is no rpm package for trixbox to download. you have to install it using .tar.gz package using ur existing linux OS. so sorry about that.
On 9/20/06, bails [EMAIL PROTECTED] wrote:
rpm -e packagename ?Rizwan
Hello ppl,
I had appdata set to use the function ENUMLOOKUP. But it gets me nothing.
| id| context | exten | priority | app |
appdata
Hi, I just set two asterisk connect with iax2 trunk.
B server
[user1]
type=user
trunk=yes
context=from-trunk
username=user1
auth=plaintext
secret=passwd
notransfer=yes
A server
register = user1:[EMAIL PROTECTED]
I notice on A's CLI, it shows Registration of 'user1' rejected:
'Registration
My client needs an option to forward the incoming calls to the ring group and to the queues to his other number when he closes his office. By default, at 9 PM the incoming calls are forwarded automatically. But he wants something so that if he closes earlier, he can forward the incoming calls
I would like to know how you got Asterisk to function with 2500 SIP
registrations. Did you have qualify enabled?
Yes, qualify was enabled, using the standard length of qualification
period between checks. Very few accounts had custom qualify settings.
What about the 500
Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card will
have two FXO and two FXS modules.
Two incoming analog lines, which need to be treated as distinct entities.
Meaning, for example, line 1= company1, line2=company2, or line 1= home line,
line2=business line. In my
So, is this GUI you speak of so often able to cater to CARRIERs rather than
ENTERPRISEs?
-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Tue 9/19/2006 10:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:
G'Day List,
Interesting article. Enjoy
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5
Mike
The text states that asterisk cannot do secretarial functions, meaning one
person, or admin, cannot
Hi All
I have 2 sip users setup in the database for realtime and they also have
their extension setup in the database.
When I register user 1 fine and can make and recieve calls.
As soon as i register user2 user1 is then unable to make any calls??
If i put the config fr both users in the
On 06:25, Wed 20 Sep 06, BJ Weschke wrote:
If you don't want such behavior, you might want to take a look at
monitoring a specific channel event in the Asterisk manager and then
starting off your script upon the receipt of such an event through the
manager.
Or if you want to go dirty:
run an
This is pound key linux from rpath. I don't see a source directory. That
is why I think I must be missing something.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, September 19, 2006 10:49 PM
To:
S McGowan,
I don't know if you missed my question (from the slew of questions you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening? Was this box only responsible for VoIP-to-VoIP calls
joea, j4computers wrote:
Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:
G'Day List,
Interesting article. Enjoy
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5
Mike
The text states that asterisk cannot do secretarial functions, meaning one person,
Hi,
I'm trying to use mysql for sip users management and i'm a bit stuck
with a problem.
I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4.
The fact is that i've put a row in the mysql sip table for my linksys
phone and i can make calls and receive calls with it, but it
I can't help you with distro specific stuff. You need MPG123 1.59r If
you do not have that version then you will experience these issues. OR
you could use the Native MOH features of Asterisk.
Jeronimo Romero wrote:
This is pound key linux from rpath. I don't see a source directory. That
is
Michiel van Baak wrote:
On 06:25, Wed 20 Sep 06, BJ Weschke wrote:
#!/bin/sh
/path/to/my/actual/script
exit 0
If you were to do that, then you might as well use System()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Is it not possible to run tx_fax over a SIP connection
to another box that dials using a TDM card?
My is seg faulting.
I can make phone calls over that same SIP connection
so everything is working there just not tx_fax?
Any idea? or is this not supported.
Jerry
Yes. It this is the opensource poundkey from rpath. I just installed madplay
instead of dealing with mpg123. Works like a charm. Is there any downside to
madplay that that I should know about??
Here my musiconhold.conf file:
-Original Message-
From: [EMAIL PROTECTED]
Hi to all.
I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and
I've some problem with outgoing calls: there is a big delay for
bidirectional audio flow.
Here is mean part of an asterisk trace releted to outgoing calls.
(canreinvite=no for both peers).
Until SIP 180 ringing
Scott Pinhorne wrote:
Hi All
I have 2 sip users setup in the database for realtime and they also have
their extension setup in the database.
When I register user 1 fine and can make and recieve calls.
As soon as i register user2 user1 is then unable to make any calls??
If i put the
Arkaitz wrote:
Hi,
I'm trying to use mysql for sip users management and i'm a bit stuck
with a problem.
I use asterisk-1.2.12.1 and res_config_mysql from asterisk-addons-1.2.4.
The fact is that i've put a row in the mysql sip table for my linksys
phone and i can make calls and receive calls
Hi,
This config is working for me:
_
musiconhold.conf
[shoutcast]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f
8192 -r 8000 http://stream128.submusic.ch:8004/
; The '/' in the stream URL
Rushowr wrote:
S McGowan,
I don't know if you missed my question (from the slew of questions you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening? Was this box only responsible for
. .
What SPECIFICALLY are you trying to do that you are unable to do?
No specifics, at this time, too early in evaluation.
I get the point, I think, about thousands of buttons.
My concerns are the ability to answer on multiple lines, and have various
options,upon no pickup,
Doug Lytle wrote:
Jamin W. Collins wrote:
callprogress = yes
The only thing I'm iffy about is the above entry.
Maybe it's mistaking the progress as disconnect?
That does appear to have been the issue. We haven't had a new
occurrence of the random disconnects since disabling callprogress.
[EMAIL PROTECTED] wrote:
Again, I'm amazed by this example since it
seems to be way over what anyone else normally reports as usable.
Exactly!
--
Kristian Kielhofner
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Hi,
Thanks, now i see the phone in show sip peers, I've been reading
about rtcachefriends and now i understand what was the problem.
But the other problem is still here :(. It seems that asterisk is
unable to find any file in the system, not gsm file nor codec...
nothing. It's strange since i
Asterisk [Submusic] wrote:
musiconhold.conf
[shoutcast]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000
http://stream128.submusic.ch:8004/
; The '/' in the stream URL is important !
I tried this.
however it doesn't work. apparently, asterisk doesn't read from
I suspect the article is referring to BLF, which is a traditional Key
System feature. It does not scale well in larger PBXs.
BLF support is not great (in Asterisk OR in phones) for SIP.
joea, j4computers wrote:
. .
What SPECIFICALLY are you trying to do that you are unable to do?
No
Jamin W. Collins wrote:
Doug Lytle wrote:
Jamin W. Collins wrote:
callprogress = yes
The only thing I'm iffy about is the above entry.
Maybe it's mistaking the progress as disconnect?
That does appear to have been the issue. We haven't had a new
occurrence of the random disconnects
Hi Sheerwood, I unfortunately saw a bit of what I percieve to be an error in what you said. BerkeleyDB does in fact support replication across nodes - see: http://www.sleepycat.com/docs/ref/rep/intro.html- possibly you meant to say the version implemented in * does not support replication. If
Eric ManxPower Wieling wrote:
The comments in /etc/asterisk/zapata.conf didn't tip you off?
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the
progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts to determine answer, busy,
magnus wrote:
Hi all, could anyone share how to perform attended transfers with Asterisk
and Grandstream SX2000's - we are able to perform blind transfers with no
problem, but attended transfers fail - is it necessary to set two line
identities on the phones to be able to do this?
Appreciate all
On Wed, Sep 20, 2006 at 03:57:07PM +0100, adebayo omo-dare wrote:
Hi Sheerwood,
I unfortunately saw a bit of what I percieve to be an error in what
you said. BerkeleyDB does in fact support replication across nodes -
see: http://www.sleepycat.com/docs/ref/rep/intro.html - possibly you
Title: Zap channel digit.
I have a problem in outbound call.
My extension is.
exten = _0X.,1,Dial(zap/1/${EXTEN},20,TW)
exten = _0X.,2,Dial(zap/2/${EXTEN},20,TW)
exten = _0X.,3,Dial(zap/3/${EXTEN},20,TW)
exten = _0X.,4,Dial(zap/4/${EXTEN\},20,TW)
exten = _0X,105,Playback(tt-allbusy)
On Fri, Sep 15, 2006 at 09:14:25AM -0500, Moises Silva wrote:
If you want to have a safe asterisk I would recommend using svscan
from daemontools package, more wonderfull software of D.J. Bernstein.
http://cr.yp.to/daemontools/svscan.html
Assumming you really want to live with DJB-style file
Setting Realm to asterisk worked for me.
ref. from sip.conf:
;realm=mydomain.tld ; Realm for digest authentication
; defaults to asterisk. If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set
I'm interested, too in how to accomplish this. I have tried earlier today
with a Snom360 to register it using its mac address as the authentication
username. I can't seem to get it to work (hopefully I'm just doing
something wrong).
My sip.conf (asterisk 1.2.12) looks something like:
Jamin W. Collins wrote:
Eric ManxPower Wieling wrote:
The comments in /etc/asterisk/zapata.conf didn't tip you off?
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the
progress
; of a call through RINGING, BUSY, and ANSWERING. If turned on, call
; progress attempts to
On Wed, 2006-09-20 at 08:26 -0500, Eric ManxPower Wieling wrote:
joea, j4computers wrote:
Ferguson, Michael[EMAIL PROTECTED] Wrote on: 9/20/2006 8:03 AM:
G'Day List,
Interesting article. Enjoy
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?t5
Mike
I'm trying to dial multiple SIP channels and check availability before I
dial them.
i.e. say I have an internal group that I define (extension 50) which
actually dials SIP extensions 51 and 53
I'd use Dial(SIP/51SIP/53), but if a phone isn't registered (i.e.
someone's unplugged 53) it does weird
On Wed, Sep 20, 2006 at 10:26:25AM +0530, Benjamin Jacob wrote:
This somewot spoils the fun in Asterisk, when talking of performance, to
query the DB for every call . Sort of pulls things down. Any comments or
observations guys?
Well, my personal observation is that if you can't make your
On Wed, Sep 20, 2006 at 04:12:34PM +0100, Faris Raouf wrote:
magnus wrote:
Hi all, could anyone share how to perform attended transfers with Asterisk
and Grandstream SX2000's - we are able to perform blind transfers with no
problem, but attended transfers fail - is it necessary to set two
Arkaitz wrote:
Hi,
Thanks, now i see the phone in show sip peers, I've been reading
about rtcachefriends and now i understand what was the problem.
But the other problem is still here :(. It seems that asterisk is
unable to find any file in the system, not gsm file nor codec...
nothing.
Hi,
I want to make a call from the box on which asterisk is run to an xlite client.How can i proceed on this what are the requirements and configurations needed.
Thanks Regards,
Saritha
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Im unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Below is my configs (Maybe I missed something)
Thanks for any help you could give!!
##sip.conf##
[general]
callerid=unavailable
context=default
; Default context for
Hi,
Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE,
India Pakistan.
Thank you.
John
mail2web - Check your email from the web at
http://mail2web.com/ .
___
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Has someone experienced the same with the FreePBX frontend?
After changing a SIP extension and pressing the red
bar on top in the browser I only see on the CLI:
sip*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Form Hold
Last Message
62.x.x.x
I have had Asterisk 1.2.10 up and running for the past two months. I
have not done anything to the system in the last month.
I am using broadvoice.com as a sip provider. Yesterday everything was
working fine and now when I call out or receive calls I cannot hear the
person on the other line,
Hi,
Looking for good rates quality.
UK mobile/landline in particular. Saudi Arabia, India, Pakistan, UAE,
Malaysia etc.
Thanks,
John
mail2web - Check your email from the web at
http://mail2web.com/ .
We are aware of the MPG123 tweaks that were always needed with Fedora
in the past. We have MOH working on all other systems.
We just installed a new system with a clean install of 1.2.12.1. It
seems that there is info on the Wiki which states that there is a new
way to do MOH using some internal
Rizwan Hisham [EMAIL PROTECTED] wrote:
If you have installed the .iso version of Trixbox then Trixbox IS your system from the
operating system (CentOS) up. The .iso wipes your disk partitions just for starters so
to revert to anything else means installing from scratch - including operating
We can do attended transfers on the GXP-2000 just fine with a single
account.
When you have a call on Line 1, simply press Line 2 (Line 1 will be
put on hold automatically) and press SEND. Once the other party picks
up, you announce the call and then press TRNSFR and then press Line 1.
-
Erik is this for a Mediatrix 1204? If so where did you get these
settings? In SNMP? or HTTP?
From the Mediatrix documentation:
Page 59 (87) These are footnotes to whereever the words register
server are mentioned in the Manual:
1. The Mediatrix 1204 does not use the Registrar server.
2. The
Hi,
this seems interesting solution... I found trysystem command too. Asap I
can I'll try them
Thank you all.
Doug Lytle wrote:
Michiel van Baak wrote:
On 06:25, Wed 20 Sep 06, BJ Weschke wrote:
#!/bin/sh
/path/to/my/actual/script
exit 0
If you were to do that, then you might as
Hi,
I'm using mpg123 to play music on hold but it seems that Asterisk does
play the music from a random point: is there a way to make my music on
hold always starting from beginning?
TIA
Giorgio Incantalupo
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You will need to change the type=friend to type=peer and
also define call-limit to some value (it can be large if you don't care about
the actual limit). That should fix hints for you.
Regards,
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall,
Eric
On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote:
I’m unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Confirmed here, I only noticed because of this message.
--
Dave Cotton [EMAIL PROTECTED]
Hi List,
Can anyone confirm if the Linksys PAP2-UK works with Asterisk. I can get
the device to register with my Asterisk box ( v1.2.12.1 ) but I don't get a
dial tone. I have no firewall on my asterisk box and all my other IP
phones work ok.
Thanks in advance.
Phil.
[EMAIL PROTECTED] wrote:
Hi,
Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE,
India Pakistan.
This is a -biz question, not -users.
Also, do you realize how bad it makes you look that you can't even
bother to put a subject on your mail?
B.
--
This message has been
Thanks for your response.
Unfortunately I still receive the same
error Error updating bootrom no matter what version of sip and the bootROM
I upload to the ftp site. I have even used the latest release of the fimware
could I have somehow broke the phone with a corrupted flash. How do
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to voicemail (which appear to need
to be g711)?
Thanks a ton!
Brian
On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Use
Tzafrir Cohen wrote:
On Tue, Sep 19, 2006 at 09:58:45PM -0700, mitcheloc wrote:
You are incorrect. The GUI you are referring to is the framework I already
mentioned. The webpages are static html javascript (AJAX functionality).
Asterisk has a simple built in HTTP server in trunk now which
Hi,
On 9/20/06, Michel Vaillancourt [EMAIL PROTECTED] wrote:
Arkaitz wrote:
Hi,
Thanks, now i see the phone in show sip peers, I've been reading
about rtcachefriends and now i understand what was the problem.
But the other problem is still here :(. It seems that asterisk is
unable to find
On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote:
You will need to change the type=friend to type=peer and also define
call-limit to some value (it can be large if you don't care about the
actual limit). That should fix hints for you.
But if you have it set to 1 then busy status
Le mercredi 20 septembre 2006 à 18:18 +0200, Giorgio Incantalupo a
écrit :
Hi,
I'm using mpg123 to play music on hold but it seems that Asterisk does
play the music from a random point: is there a way to make my music on
hold always starting from beginning?
Use native format audio (ulaw,
Just found out this may only been a sip problem.
State work with zap and SCCP when checking status via cli
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton
Sent: Wednesday, September 20, 2006 12:31 PM
To: Asterisk Users Mailing List -
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