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
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNS
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 a
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:
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}
/var/spool/asterisk/out
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 never,
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 an
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 voice
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
resellers
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: http:
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 phon
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
enc
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 S
I'm confused about
SIP realtime updates. If I make a database change, and then do a "sip prune
realtime 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",
Asterisk q
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 quest
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 AT&T/SBC.
AT&T/SBC is pretty much a st
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 va
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 re-reg
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,
(p
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 h
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 AT&T/SBC. Thanks Julian
___
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 somethin
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 i
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
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 guaran
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 th
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
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:
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,
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
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
___
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 '/etc/asterisk/sip_no
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)
Ar
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
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 A
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
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 SVN-branch-1.
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 k
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 incorp
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...
An
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:
>
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 4BR
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 Mai
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 to
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 al
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
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 ...
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 worka
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
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
> y
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 o
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> --
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 xxx.xxx.x
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 melw
> 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, Ark
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
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 i
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 Coins
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
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 <[EM
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 foll
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
aster
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.
Th
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 my
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
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
http://www.voiceone.it/forum/viewforum.php?f=
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, in
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 'de
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 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 ctx:
>>
>> 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
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:
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
I am surprised that you are getting echo on SIP
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 re
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 écrit
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 this
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 gene
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 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 da
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
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]
type=
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 stol
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 t
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
username=from_serv
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
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
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 "iax.renoir
On Fri, 27 Oct 2006, Michiel van Baak wrote:
> On 23:11, Thu 26 Oct 06, Armin Schindler wrote:
>
> > 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 i
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
___
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
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
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 e
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