Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my
update of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for example if i call
myself via
Maybe a missing space between expr1 and the sign on extension s
priority 3 ?
[]'s
MM
-Original Message-
From: Esteban Guana-Jarrin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 27 Oct 2006 15:11:33 +1000
Delivered: Fri, 27 Oct 2006 01:51:20
Try this:
exten = s,3,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?7:4)
exten = s,4,Gotoif($[$[${test11} = yes] $[${test12} = no]]?9:5)
exten = s,5,Gotoif($[$[${test11} = no] $[${test12} = no]]?11:6)
exten = s,6,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?13:15)
A $[] for the entire
On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote:
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
Since they are incorporated in a single product which is doing the
configuration, consistency where possible would be good...
That product is designed to link the two
MM,
The $[] made it work. Thanks a lot for your assistance.
obligado :)
See the debug output below.
-- Executing NoOp(SIP/123-d14f, no) in new stack
-- Executing NoOp(SIP/123-d14f, yes) in new stack
-- Executing GotoIf(SIP/123-d14f, 0?7:4) in new stack
-- Goto (test-check,s,4)
--
They aren't zap interfaces unfortunately. They are SIP/IAX channels
started from originate and the manager API.
Lenz wrote:
why not using a zap show command and parse the results externally?
l.
On Thu, 26 Oct 2006 13:12:46 +0200, Nick Adams [EMAIL PROTECTED] wrote:
I need to determine the
On 2006-10-26 23:02:40 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no
I've enabled those options but it's the same.
On 10/25/06, Maxi Belino [EMAIL PROTECTED] wrote:
i'm having similar problems (if you find out the solution please post it)
did you try enabling 'callprogress' or 'busydetect' in zapata.conf ?
Maxi
2006/10/23, Arkaitz [EMAIL PROTECTED]:
Hi,
Im
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
For my home Asterisk setup I have a single PSTN line, and then I use a
variety of different voip providers. I use two different providers for
my DID's (one toll free, and one normal). I use yet a different provider
for terminating
Hi guys,
is there a comment from digium on the license of chan_skype? I could not
find the GPL_KEY in the precompiled module, and they don't release the
source. So i'm guessing, they'd need a commercial license...?
Regards,
Andreas
On Fri, 27 Oct 2006, Michiel van Baak wrote:
On 23:11, Thu 26 Oct 06, Armin Schindler wrote:
snip/snip
chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi
with more features and as far as I can tell, much more stable.
You do faxing with chan-capi 0.3.5? But this
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include
Hi.
I have now many customers using hylafax + asterisk, and all of
them have proven to be reliable.
Two are using diva server 4bri 8m + CAPI (asterisk here is not
involved, as the incoming fax call gets directly to ttyds0x devices,
and the numbers assigned to the lines by our telco are excluded
On Fri, 27 Oct 2006, Thomas Winter wrote:
Am Thursday 26 October 2006 23:35 schrieben Sie:
On Thu, 26 Oct 2006, Thomas Winter wrote:
I would recommend the Eicon DIVA Server 4BRI cards. They have a
capi interface which is used by chan-capi (chan-capi.org) and
onboards DSPs for the faxing.
Hello list,
I try to configure auto dial from asterisk (called server B) to another
asterisk (server A) using SIP but I have a strange problem !
(Softphone connected to server B calling clients of server A works)
On server B, I have :
sip.conf :
[to_serverA]
type=peer
Hi list,
I want to make a statistics about the number of parallel calls on my *
running a beronet E1 card. The easy variant would be to get a number of
maximal parallel calls to my machine during a day. The extended would be
a graph showing the load over the day.
If noone knows a direct solution
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
Hi,It seems to me that Bristuff usage has reached a point which implies a dedicated mailing list.This list would be of major use for :- bugs assessment- features requests- comments on Asterisk news
Who seconds that ?Would it be difficult to make this happen ?Regards
On Fri, Oct 27, 2006 at 07:56:16AM +0800, Leo Ann Boon wrote:
Moises Silva wrote:
AFAIK, you will need to do the first. ARA-odbc-sqlite
res_sqlite3 in asterisk-addons supports ARA
res_sqlite3 from aadd-ons is a strange beast. It uses its own, private
copy of sqlite and acceses internal data
2006/10/27, Armin Schindler [EMAIL PROTECTED]:
On Fri, 27 Oct 2006, Thomas Winter wrote: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a
capi interface which is used by chan-capi
On Thu, Oct 26, 2006 at 04:17:16PM -0700, Alyed Tzompa wrote:
Echo is generated by the analog end to where you place the
call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in
the zapata.conf (since IP calls don't generate
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with
Thanks to take time to write me back (oola I' don't no if this is a
correct sentence !)
I think the variable sevret is empty is not a problem : without it it's
the same !
I will try to debug with type=peer and type=user
I didn't know this site, hope it will be helpfull !
jb
Marco Mouta a
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk
Message: 7
Date: Thu, 26 Oct 2006 22:56:58 -0400
From: Michael Araba [EMAIL PROTECTED]
Subject: [asterisk-users] RE: ECHO Cancellation in SIP Calls
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
I am surprised that you
Any experts on porting numbers in the uk here? ;-)
Yep, it is your legal _right_ to have the numbers ported in a
reasonable time/cost.
Point this out to them and ask what the complaints escalation
procedure is. That should get their attention.
Can you point me to the law that gives you the
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.
P[ 1] * Starting Ast
Hello all
Asterisk 1.2.10
BRIstuff PRE-1s
Debian sarge
I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is
down, no matter is I have an actual line plugged in or not.
== Primary D-Channel on span 1 down
I somehow got it to work once! The config tools do not indicate any
Hi for all,
i 've installed asterisk with isdn trunk with alcatel pabx.
For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour:
!! Unexpected Channel selection 3
-- Extension '' in context
Hi for all,
i 've installed asterisk with isdn trunk with alcatel pabx.
When alcatel users dial for external numbers, a channel on asterisk is
allocated for dial. When we call to an number that is an IVR the digits
isn't recognized by IVR.
In sip.conf i putted dtmfmode as rfc... and info,
Alex ha scritto:
Hi all!
We've released VoiceOne 0.4.0, a web-based and open source solution
which allows to fully manage an Asterisk service hosted on a LAMP server.
Thanks guys, translators and testers are welcome!
We have a dedicated forum at
Hi Mark,
why
exten = *kpn-in*,1,Dial(SIP/mark,25,tr) ??
Try:
exten = s,1,Dial(SIP/mark,25,tr)
and
exten = _X.,1,Dial(SIP/mark,25,tr)
Giorgio Incantalupo
Mark Hannessen wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't
I am taking a Polycom IP601 home to try to figure out how to provision
it outside of the office for our outsides sales people.
Our asterisk server has a direct outside IP.
The IP601 will be behind a router at home so it will not have an outside IP.
I am fully opening the company firewall for
Thanks. I will give that a try. Do you know if removing that line will
affect
other phones I might have?
If so, maybe I am better off getting someone else's phone.
ACT's support seems a bit problematic. They responded to my first email right
away,
but never, so far, to my second.
pls post your misdn.conf as well as extensions.conf, so someone could
help you on this.
On 10/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:
Hi for all,
i 've installed asterisk with isdn trunk with alcatel pabx.
For alcatel users use asterisk lines, should dial 0 to take tone from
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the
My mistake:
[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL
If you are calling from a SIP phone through asterisk and through a Digium card,
one could argue that the Digium card IS farside of
the SIP phone.
SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN --
Destination.
I would argue that the Digium card IS on the farside of asterisk
Check your dtmfmode
I use dtmfmode=rfc2833
Check with your provider
Best regards,
Al Bochter
Bochter Services
(Voip PBX) Toll Free: 866-638-1254 EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250
(Voip) Cellular: 712-432-5401
http://www.BochterServices.com/?t=Email
BUY and sell
Frédéric Blaise ha scritto:
Hello all
Asterisk 1.2.10
BRIstuff PRE-1s
Debian sarge
I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is
down, no matter is I have an actual line plugged in or not.
== Primary D-Channel on span 1 down
Try with signalling=bri_cpe even
Hi Frederico,
I had digits detection problems with my ISDN beronet cards too, do not
know if u are using those cards but in case try to add s parameter to
Dial command:
dial(mISDN/1/123/s)
It worked for me. :)
Giorgio Incantalupo
Frederico Madeira wrote:
Hi for all,
i 've installed
I've enabled those options but it's the same.
On 10/25/06, Maxi Belino [EMAIL PROTECTED] wrote:
i'm having similar problems (if you find out the solution please post
it)
did you try enabling 'callprogress' or 'busydetect' in zapata.conf ?
Maxi
2006/10/23, Arkaitz [EMAIL
Olivier ha scritto:
What about telephony features using chan-capi and Asterisk ?
Are those features on par with msidn+Asterisk or bristuff+Asterisk
(maybe I'm mixing up things together) ?
Cheers
I'm running my own company's pbx with diva 4bri, diva server for linux 8.2,
chan_capi from
Hi people,
pls does anybody know what (T) and (D) letter means?
server3*CLI iax2 show peers
Name/UsernameHost Mask Port Status
SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK
(29 ms)
SERVER2
Hello Users,
Good Morning,
In Conferemcing How to Disconnect the phone while in between the
Conference .
When I press the ' # ' key for Disconnecting the
Conference..
Below the Following to shows some Warning, ( in Red Color )
from-sip en
*CLI -- Executing
Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look
On Fri, 2006-10-27 at 14:14 +0200, Alberto Pastore wrote:
Try with signalling=bri_cpe even if your lines are
Yes, I tried with all kind of signalling, including this one, but this
doesn't work either. bri_cpe_ptmp seems to be the one...
set as point to multipoint, at least that should make
Hello all
I would like to know your opinions on free GUI used to manage Asterisk.
Which is better?
My setup is quite small, about 15-20 phones. I've seen the liste on
voip-info.
Thanks all.
fred
signature.asc
Description: This is a digitally signed message part
I have a bunch of Snom phones. When I press the mute button, the phone
stops sending RTP frames. If I have rtptimeout set, that means that
the connection will eventually be cut off. It also affects sound
generated by asterisk, since timing is generated from the incoming
frames.
Are there any
Why r u using rsa authentication? you should start with something
simple. test the link i sent u.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
In the source that I've read (admittedly it's pretty old - 1.2.7.1)
SipAddHeader() only appears to work on INVITEs.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 26 October 2006 23:23
To: asterisk-users@lists.digium.com
issue #8126 (http://bugs.digium.com/view.php?id=8216) on mantis is a
patch for the queue system which allows you to specify a macro to run
when a member is connected to a queue call, either by a configuration
parameter in queues.conf or as an optional parameter on the Queue
application.
It
hi, does anyone know if it is possible to set the outgoing msn number with
chan_misdn (the number the people on the other side will see as the caller)
I already tried
Set(CALLERID(num)=1234)
SetVar(CALLERIDNUM=1234)
Set(CALLERID(name)=1234[|a])
Set(CALLERID(number)=1234)
but none of them seem
Make sure you set nat=yes for the sip user. Asterisk will then send replies
back to the source IP address, rather than what's in the Via: header.
-Original Message-
From: Warren (mailing lists) [mailto:[EMAIL PROTECTED]
Sent: Friday, October 27, 2006 5:45 AM
To: Asterisk Users
On Fri, 27 Oct 2006, Olivier wrote:
2006/10/27, Armin Schindler [EMAIL PROTECTED]:
On Fri, 27 Oct 2006, Thomas Winter wrote:
Am Thursday 26 October 2006 23:35 schrieben Sie:
On Thu, 26 Oct 2006, Thomas Winter wrote:
I would recommend the Eicon DIVA Server 4BRI cards. They have
Cohen, so you vote for the ARA-odbc-sqlite route?
this is for embedded, so that's why sqlite instead of mysql or postgres.
when you say it is not guaranteed, what do you mean?
On 10/27/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Oct 27, 2006 at 07:56:16AM +0800, Leo Ann Boon wrote:
Hello,
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but
I'd really like to understand what's going on there...
Hi
Which is most resistant to the loss of packages in a dirty link ? SIP or IAX ?
roberto
2006/10/27, Dave Cotton [EMAIL PROTECTED]:
On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote:
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
Since they are
Hello everyone.
I know it's a little bit off-topic, but I was just wondering...
Has anyone ever had any experience with asterisk,
a wi-fi meshed lan (with more than one access point)
and wi-fi sip phones?
I made some tests but I'm not really satisfied
Wi-fi phones are a curse (as far as I
Louis-David Mitterrand wrote:
Hello,
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
I'm running just 2.6.18 fine Under 1.2 Branch without issue.
Connected to Asterisk
On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote:
Louis-David Mitterrand wrote:
Hello,
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
I'm running just 2.6.18 fine
Hi all,
I have problems receiving calls from PSTN with an Wildcard T207P.
All internal SIP devices have a 3 digit extension, e.g. 873.
When I call the extension from the PSTN this way everything works fine:
1. enter the number on the phone
2. lift off the handset
But when I call it that way
On Fri, Oct 27, 2006 at 05:11:24PM +0200, Louis-David Mitterrand wrote:
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
Downgrading back to latest 2.6.17.x solves the problem for
Louis-David Mitterrand wrote:
On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote:
Louis-David Mitterrand wrote:
I'm running just 2.6.18 fine Under 1.2 Branch without issue.
Connected to Asterisk SVN-branch-1.2-r44580 currently running on livonia
(pid = 7349)
Please help
I am using [EMAIL PROTECTED] 2.6
Since I enter the conference prompt its will ask for the
password ,after that it said invalid conference number
Remark the password is correct but it cant know that it have
a conference number (555)
== Parsing
Can you be more specific? What sort of linkages are available between
the two offices?
CP
On 22-Oct-06, at 10:38 PM, dthurn wrote:
What's the best way to connect an Asterisk PBX to a Nortel MICS PBX.
I have two offices that I want to link together.
TTFN
Hello;
I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac
Pro intel box.
When I try to record a message from an incoming call or a greeting
message from internal phone using voicemail, It's like something is
not doing well. I can heard anything, only a distorsion sound that is
You might check with Aastra, they are showing a DECT phone that will work with Asterisk via sip. I know the release is next year for me, but since you are in Europe it may be avaliable sooner.
On 10/27/06, Alberto Pastore [EMAIL PROTECTED] wrote:
Hello everyone.I know it's a little bit off-topic,
I'm already try this configuration, but don't have sucess.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br
2006/10/27, Al Bochter [EMAIL PROTECTED]:
Check your dtmfmode
I use dtmfmode=rfc2833
Check with your provider
Best regards,Al BochterBochter Services(Voip PBX) Toll Free:
On Fri, Oct 27, 2006 at 08:10:30AM -0400, Steven wrote:
If you are calling from a SIP phone through asterisk and through a Digium
card, one could argue that the Digium card IS farside of
the SIP phone.
SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN --
Destination.
If
We have a large number of numbers (!) that we need to clean from our
database. I've been asked if we can do this automatically, by checking
if the number is valid or not from asterisk.
what I don't want to do is to disturb the phone owners if the number is
valid.
obviously I can catch all
On Fri, Oct 27, 2006 at 09:56:09AM -0500, Erick Perez wrote:
Cohen, so you vote for the ARA-odbc-sqlite route?
Can't think of anything better, now. But I haven't actually tried using
it.
this is for embedded, so that's why sqlite instead of mysql or postgres.
when you say it is not
On Fri, Oct 27, 2006 at 04:35:52PM +0200, Armin Schindler wrote:
On Fri, 27 Oct 2006, Olivier wrote:
2006/10/27, Armin Schindler [EMAIL PROTECTED]:
On Fri, 27 Oct 2006, Thomas Winter wrote:
Am Thursday 26 October 2006 23:35 schrieben Sie:
On Thu, 26 Oct 2006, Thomas Winter
On Fri, Oct 27, 2006 at 02:14:43PM +0200, Alberto Pastore wrote:
Frédéric Blaise ha scritto:
Hello all
Asterisk 1.2.10
BRIstuff PRE-1s
Debian sarge
I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is
down, no matter is I have an actual line plugged in or not.
On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said:
Hello;
I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac
Pro intel box.
When I try to record a message from an incoming call or a greeting
message from internal phone using voicemail, It's like something
Hi Groupies,I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC.ThanksJulian
___
On 2006-10-27 08:49:44 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Hello everyone.
I know it's a little bit off-topic, but I was just wondering...
Has anyone ever had any experience with asterisk,
a wi-fi meshed lan (with more than one access point)
and wi-fi sip phones?
I don't think I
Hi,
My users are currently using a console interface like this:
see it at: http://www.whssf.org/interface.jpg
which came with a Praxon PDX we got about 5 years ago, which is now
unsupported,
it works very good and converts any analog phone plugged into the
system into a powerful console,
Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but
I think the biggest issue with with telemarketers. I get blatently illegal calls all the time, besides the fact that I am on the do not call lists. Today I got a call from some group trying to sell me a Razr phone for $50, automated computer, no option to remove yourself and the callerid appears
Julian Varanini wrote:
Hi Groupies,
I am sort of new to the whole asterisk thing, especially when it comes
to the Digium TE110P card. Does anyone have experience setting this
up? If so can you help me out? The provider for the PRI is going to
be ATT/SBC.
ATT/SBC is pretty much a
Hello,
I am currently running 1.4-Beta3 on my test system and have enabled the new
HTTP functionality. I have enabled http and web in both http.conf and
manager.conf. I can succefuly reach:
http://localhost:8088/asterisk/httpstatus
http://localhost:8088/asterisk/static/ajamdemo.html
My
I'm confused about
SIP realtime updates. If I make a database change, and then do a "sip prune
realtime peer peer", I can see Asterisk query the database, and retrieve
the updated information. However, it still uses the old values. What's up with
that?
If I do a "reload",
On Fri, 27 Oct 2006 13:38:51 -0500, LJ wrote
Hello,
I am currently running 1.4-Beta3 on my test system and have enabled
the new HTTP functionality. I have enabled http and web in both
http.conf and manager.conf. I can succefuly reach:
You have to download it manually from SVN as
I've got a Zultys WIP2 and Zultys 2x2 both of which support encryption. I
have patched my asterisk with srtp (srtp.sourceforce.net) as well as with
the patch found at http://bugs.digium.com/view.php?id=5413. I'm trying to
utilizing the encryption feature of the two Zultys phones to create an
I am not familiar with the SNOM phone. On some mfg phones I think they have
a setting to enable transmit silence. See if Snom has such a setting.
Benny Amorsen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
I have a bunch of Snom phones. When I press the mute button, the phone
Greetings,
This is my annual post-Astricon attempt to start an Asterisk User
Group in the Vancouver, BC, area. If you are interested, please reply
off-list.
Regards,
--
Anthony Rodgers (CunningPike)
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed:
Greetings,
This is my annual post-Astricon attempt to get an Enterprise Asterisk
User Group off the ground. We are a municipal government using
Asterisk to replace a legacy PBX. I'd be interested in starting a
group of similar enterprise users (say, 100 seats or more) other than
This seems to be a bug.
I can get exitcontext to work on a per mailbox basis in voicemail.conf.
However, for realtime mailboxes, I added a new column called 'exitcontext' to
my table, and the thing simply doesn't work. I can see asterisk selecting *
from the table, but pressing 0 while in
In my setup, sip calls coming in through a proxy with a sip.conf entry set to autocreatepeer=yes and context=proxy get placed into context proxy in the dial plan. That is expected.However, if the username in the From: address exists in the sippeers table, it gets challenged for the password and on
joe, at j4computers wrote:
Thanks. I will give that a try. Do you know if removing that line will affect
other phones I might have?
If so, maybe I am better off getting someone else's phone.
ACT's support seems a bit problematic. They responded to my first email right away,
but
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
Hi,
Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo)
Steps:
modprobe zaptel
modprobe wctdm
ztcfg -vv
/etc/zaptel.conf
fxsls=1-4 # TDM04B
defaultzone=us
loadzone=us
/etc/asterisk/zapata.conf
signalling=fxs_ls
group=1
context=incoming
channel = 1-4
modprobe zaptel
hello,
is it possible to dial out external number within running conference,
for example dial out using zap channel and connect to pstn conference,
thx
bart
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