On Mon, 2009-12-14 at 22:27 +0200, Tzafrir Cohen wrote:
> On Mon, Dec 14, 2009 at 02:17:39PM -0300, Vinícius Fontes wrote:
> > I have never used that card myself, but I have never seen an analog
> > board reporting a RED alarm.
>
> Ahem. Wcfxo always has (AFAIR). "Red alarm" means that no line is
On Mon, 2009-12-14 at 14:10 -0600, Tilghman Lesher wrote:
> > I don't want to start a war, but there is a square to that. I'm new to
> > Asterisk having spent years in analogue telephony. If I can get a test
> > Asterisk working on a cheap clone card without a hitch, I'm most likely
> > to expand
IAXDIAL is free on app store works great on WiFi even true NATs but seem
blocked for GPRS.
HB
>>> Re: [asterisk-users] iphone client app
>>> From:
>>> Alex Samad
>>> Date:
>>> Tue, 15 Dec 2009 12:08:37 +1100
>>> To:
>>> asterisk-users@lists.digium.com
>>>
>>> To:
>>> asterisk-users@lists.digium.
Hi All;
When using the digest authentication method, so I have to create the realm
domain with its username and passwords to be used for SIP digest
authentication, correct?
Now, how to create this domain? Should be reachable (can be ping) from a remote
device?
In other words, to create this
Joseph,
You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
provides real FXO port that registers with Asterisk.
David
On Sat, Dec 12, 2009 at 1:37 AM, Joseph wrote:
> I'm looking for a reliable ATA FXO/FXS adapter.
>
> Linksys 3102 - a lot of echo problem + two of them died wit
> this i got from syslog:
>
> puppy:~# grep pulse /var/log/syslog | tail -3
> Dec 14 20:32:45 puppy pulseaudio[25967]: main.c: Unable to contact D-Bus:
> org.freedesktop.DBus.Error.Spawn.ExecFailed: /usr/bin/dbus-launch
> terminated abnormally without any error message
> Dec 14 20:32:46 puppy puls
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to
learn how asterisk comunicates from server to server. I already have a server
running smoothly now, I'm installing another one to test it along side the
actual one.
I would like to run different scenarios:
Well I have a 3gs - will tell you how that goes.
decided against siax - have to pay for the base model.
installed fringe, but no voip over 3g, have to wait till i get home, but
it registered with my asterisk server so ..
I am looking for the hacked fring.ipa which allows voip over 3g, just so
I
Pat Fleet, the original voice of AT&T recorded a free set of the
prompts included in Asterisk and also does custom IVR prompts through
her website at http://patfleet.com/ I'm not sure what the going rates
for IVR prompts is, but she charges $15/phrase.
On 12/14/09, Barry L. Kline wrote:
> -BE
I find that Siphone works great on the iTouch. Tried it with my own
asterisk box as well as Callcentric and MagicJack and it was very
clear and stable. Haven't played with it since the last firmware
update though as the update removed support for 3rd party headsets .
On 12/14/09, Alex Balashov wr
--- On Mon, 12/14/09, Vieri wrote:
> From: Vieri
> Subject: Re: [asterisk-users] Asterisk ZAP/DAHDI reads phantom digit on
> overlap PRI
> To: asterisk-users@lists.digium.com
> Date: Monday, December 14, 2009, 3:26 PM
>
> --- On Mon, 12/14/09, Tzafrir Cohen
> wrote:
>
> > On Mon, Dec 14, 20
I personally have not had much luck with these softphones because the
iPhone 3G seems to be underpowered and just doesn't run them well
enough to sustain good voice quality, irrespective of wifi network
conditions. I could be mistaken, though.
It's not going to happen over AT&T's 3G network
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote:
> Fring, it's free and works perfectly with an Asterisk server..
thanks
>
>
> On 13 Dec 2009, at 10:15, Alex Samad wrote:
>
> > Hi
> >
> > Got a new iphone, want to know about peoples experience with any apps
> > that work we
When agents are on the phone, and the CLI queue show command shows their
status as busy, the queue still tries to send them calls.
Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add
agents. ringinuse is set to no for queue. Agents are using Polycom 430s.
dialplan:
exten => s
Does anyone know a sip client that can be installed on Nokia /Symbian that
register to asterisk directly , i installed Fring ,seems that the register
goes to an intermediate server on Internet that forward it to my asterisk
server .
--
Girgis Rasmy
___
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Gibbons wrote:
> This may belong on -biz, but does anyone have experience with a decent and
> cheap IVR/prompt recording house?
>
> Are decent and cheap mutually exclusive?
>
> A nice *sounding* lady would be nice... you can keep any burly voi
On Monday 14 December 2009 03:20:11 pm Stephen Davies wrote:
> On 12/14/09, Lenz Emilitri wrote:
> > But more dynamical, so I would try and look up the actual channel in the
> > AstDB, like:
> >
> > exten => XXX,hint,${DB(myagent/${EXTEN})}
> >
> > This does not seem to be working - is there a way
Thanks Victor and Vinícius for the information.
I will not be doing any transcoding but using some AGI scripts, I will
update the status once I configure and start using them.
Thanks
Sandesh
On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor
wrote:
> Hi!
>
> Having two TE410P with heavy load in
This may belong on -biz, but does anyone have experience with a decent and
cheap IVR/prompt recording house?
Are decent and cheap mutually exclusive?
A nice *sounding* lady would be nice... you can keep any burly voice studios to
yourself :)
Thanks
Dave
___
What you are missing is the new state-interface parameter to AddQueueMember.
You can't use functions in a hint exten.
Steve
On 12/14/09, Lenz Emilitri wrote:
> Hello all,
> I am trying to set up a dynamic channel to be used as an Agent dialer for a
> queue - you know, trying to replace AgentCal
Hello all,
I am trying to use asterisk 1.4.24 so that I can get app_rxfax working, I
installed it, along with the versions of libpri and zaptel that had release
dates closest to the release date of 1.4.24, however, I now have a problem
where outbound dialing now fails, cause 99 on the PRI.
Doe
Hello,
I would like to connect 2 asterisk boxes together, so this is my scenario:
Asterisk Main: it is connected to many sip providers and its main purpose as
a call termination forwarder.
Asterisk B: its connected to E1, and its purpose to terminate calls. It
will receive SIP messages
Again, more info:
Since I added rtcachefriends=yes this problem went away, but I don't really
want the friends to be cached, because I want changed to be applied ASAP.
Does anyone else have experience of the peers being unregistered before their
time with rtcachefriends=no?
Nic.
From: asteris
On Mon, Dec 14, 2009 at 02:17:39PM -0300, Vinícius Fontes wrote:
> I have never used that card myself, but I have never seen an analog
> board reporting a RED alarm.
Ahem. Wcfxo always has (AFAIR). "Red alarm" means that no line is
connected (it gets no current from the remote FXS in the central
--- On Mon, 12/14/09, Tzafrir Cohen wrote:
> On Mon, Dec 14, 2009 at 05:32:21AM
> -0800, Vieri wrote:
> > Hi,
> >
> > I've noticed that a small but meaningful quota of
> calls from my Alcatel PBX to Asterisk are failing.
> >
> > This does not always happen and it is not easily
> reproducible b
On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
> Hi there,
>
> I just upgraded a relatively old Asterisk installation (1.2) in our
> office to a relatively new version (1.6svn from last wednesday) which
> runs a Junghans QuadBRI card [1].
>
> To get this flying I got dahdi-li
On Mon, 14 Dec 2009 11:02:02 -0800, Dave Platt
wrote:
>>> See if it plays back properly.
>>
>> Running aplay as asterisk user seems to be no problem:
>>
>> aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
>> Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
>>
On Mon, Dec 14, 2009 at 05:53:40PM +0200, gurel kaynak wrote:
> Hello,
>
> I've been trying to setup asterisk with zaptel for the last 3-4 days. I had
> a lot of errors and fixed all of them but asterisk still didn't work. Then I
> saw that zaptel couldn't be loaded because I was on a vserver and
On Monday 14 December 2009 12:52:50 pm listu...@spamomania.co.uk wrote:
> On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote:
> > On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
> > > I've spent a week playing with Asterisk 1.6 and I love it. What a
> > > brilliant pie
Hi,
I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here
my settings:
chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
facilityenable=yes
rxwink=300 ; Atlas seems to use long (250ms) winks
; w
I'm just curious to know if anyone is using a usb 2.0 / ISDN30
(specifically EuroISDN) device. We are looking to purchase another pci
card, but was wondering if anyone has any horror / success stories to
share regarding a usb device.
TIA
Julian
___
--
Hi!
Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh :
> Hi,
> I was able to implement T122p one port PRI and was able to call out, but I
> am planning to use T
The easiest solution to deal with this is to have one context with
different extensions for the different numbers and route the incoming
calls from there. It should look something like this (not a tested
piece of asterisk script, just an example to give the idea).
Hope it helps :-)
Erik de
On Mon, 14 Dec 2009 11:02:02 -0800, Dave Platt
wrote:
>>> See if it plays back properly.
>>
>> Running aplay as asterisk user seems to be no problem:
>>
>> aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
>> Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
>>
Hi,
does anybody know how I can make this two configuration-settings with
Asterisk-iaxmodem for ourtgoing faxes with Hylafax on an ISDN-line?
1. disconnect-time after dialing without an answer. (is now 30 seconds, but it
must be higher)
2. isdn-service set to "3,1 kHz Audio", (is now "speech")
On 12/14/2009 06:45 PM, Olivier wrote:
>
>
> 2009/12/14 Christian Theune mailto:c...@gocept.com>>
>
> Hi there,
>
> I just upgraded a relatively old Asterisk installation (1.2) in our
> office to a relatively new version (1.6svn from last wednesday) which
> runs a Junghans QuadBRI c
Your dahdi/system.conf seems fine. But your chan_dahdi.conf is missing some
lines.
[trunkgroups]
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbrid
On Wed, Nov 11, 2009 at 8:02 PM, Kevin P. Fleming wrote:
> Jonathan Thurman wrote:
>
> > Any chance that 64 bit Linux will be supported?
>
> There is a small chance; I've done some work in the past week while
> traveling to attempt solve the 64-bit problems, and I fixed some of them
> but not all
>> See if it plays back properly.
>
> Running aplay as asterisk user seems to be no problem:
>
> aster...@puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav
> Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit
> Little Endian, Rate: 48000 Hz, mono
> aster...@puppy:~$ aplay
On Mon, 2009-12-14 at 11:11 -0600, Tilghman Lesher wrote:
> On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
> > I've spent a week playing with Asterisk 1.6 and I love it. What a
> > brilliant piece of software!
> >
> > Progress and learning have been reasonably good. I have
On Mon, 2009-12-14 at 14:17 -0300, Vinícius Fontes wrote:
> I have never used that card myself, but I have never seen an analog board
> reporting a RED alarm. Probably there is something incorrect in your
> configuration. Please post your /etc/dahdi/system.conf and
> /etc/asterisk/chan_dahdi.con
The calls itselves doesn't take a lot of CPU resources, even more considering
you're willing to use hardware echo cancelling. The real CPU hogs are apps like
MeetMe() and AGI scripts. Those are no worse than audiotranscoding thought.
You also should design the system in such way there's as few t
Hello all,
I am trying to set up a dynamic channel to be used as an Agent dialer for a
queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
I would like to do something like:
[myagents]
exten => XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
exten => XXX,n,Dial(${realchan},tT,
Hi!
Trying to figure out how to rewrite calling number of an incoming call...
A cell phone (0733025975) dials a X-Lite (977).
X-Lite "shows" 733025975 at the display, but I want it to be 0317998975.
I thought i could do something like:
exten => 977/733025975,1,Set(CALLERID(number)=0317998975)
ex
On Tuesday 24 November 2009 13:10:43 Tzafrir Cohen wrote:
> On Tue, Nov 24, 2009 at 01:07:28PM +0100, Eckhard Jokisch wrote:
> > Hi,
> > I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
> > minutes/month for free.
> > As I understand asterisk will pick the first available li
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor a
2009/12/14 Christian Theune
> Hi there,
>
> I just upgraded a relatively old Asterisk installation (1.2) in our
> office to a relatively new version (1.6svn from last wednesday) which
> runs a Junghans QuadBRI card [1].
>
> To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
This has to be easy, but I have spent a fair amount of time looking for
a solution to no avail. I am trying to get multiple phones to ring when
a call comes into an Asterisk box from a particular phone number. What
happens is that only one of the phones rings.
I have several GrandStream BT200 SI
I have never used that card myself, but I have never seen an analog board
reporting a RED alarm. Probably there is something incorrect in your
configuration. Please post your /etc/dahdi/system.conf and
/etc/asterisk/chan_dahdi.conf.
Vinícius Fontes
www.asteriskforum.com.br - Informações e dis
On Monday 14 December 2009 10:28:08 am listu...@spamomania.co.uk wrote:
> I've spent a week playing with Asterisk 1.6 and I love it. What a
> brilliant piece of software!
>
> Progress and learning have been reasonably good. I have external SIP
> provider calls coming in and have put together a litt
Hi there,
I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].
To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
well.
After a while of juggling it "wo
The SDP response is the IP of the Trixbox server.
On Fri, Dec 11, 2009 at 1:38 PM, Christian Victor
wrote:
> Hi!
>
> Are you sure you are getting Astrisk out of the media path? I guess
> reinvite must be allowed. Then it should work without transcoding
> licenses.
>
> Maybe you should take a look
I've spent a week playing with Asterisk 1.6 and I love it. What a
brilliant piece of software!
Progress and learning have been reasonably good. I have external SIP
provider calls coming in and have put together a little call platform
and I'm stunned at the flexibility.
There is one issue for me.
Hello,
I've been trying to setup asterisk with zaptel for the last 3-4 days. I had
a lot of errors and fixed all of them but asterisk still didn't work. Then I
saw that zaptel couldn't be loaded because I was on a vserver and I didn't
have the devices under /dev/zap/. I asked the system guys to in
I have tried this with windows firewall both on and off - same problem.
Thanks,
Nic.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: 14 December 2009 14:53
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] As
Are you sure this isn't a Windows zeroconfig problem? If Win drops the
connection while * is talking to your client, the registration could drop
too..
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: Monday,
Bump! And some more information (see below for initial problem):
This problem is intermittent, but you don't have to wait long for it to happen.
Also, sometimes when the reregister happens (and the client has been wrongly
unregistered) asterisk sends the correct response to the client indicating
> I'm using asterisk meetme function like:
>
>exten => 9070,n,MeetMe(|dcM)
>
> and everything works pretty well. But I would like to add a review of
> the entered conference number before the user jumps into the conference.
>
> Somthing like:
> *:"Please enter the conference number followed
On Mon, Dec 14, 2009 at 05:32:21AM -0800, Vieri wrote:
> Hi,
>
> I've noticed that a small but meaningful quota of calls from my Alcatel PBX
> to Asterisk are failing.
>
> This does not always happen and it is not easily reproducible but on high
> traffic I do get a large number of cases.
>
>
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last "incoming" label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
Your provider is probably sending the DID in the SIP header TO: field. Th
Hi there,
I'm using asterisk meetme function like:
exten => 9070,n,MeetMe(|dcM)
and everything works pretty well. But I would like to add a review of
the entered conference number before the user jumps into the conference.
Somthing like:
*:"Please enter the conference number followed by the
Daniel Stefanus wrote:
> Hi,
> I want to reconfigure my asterisk dialplan.I have a problem.I have 4
> agents in a queue.How is the configuration for the asterisk dialplan if
> I want to have only 4 agents maximum who can receive the phone,so if the
> fifth caller try to entering the queue they will
Did move 0317998975 phone from my home to my office, didnt need any:
nat=yes in sip.conf, everything worked.
I did also add callcounter=yes in sip.conf so I am not sure how it
will work when I move the phone to my home and need nat=yes again.
Will do some tests later tonight when I am at home.
On
Thx!
Did try "callcounter=yes" and it worked the way u told me!
It might have solved another problem 2, need to do some more tests...
On Sun, 13 Dec 2009 15:14:22 -0500, Leif Madsen wrote:
Philipp Kempgen wrote:
> Magnus Benngård schrieb:
> Set
> call-limit=10
> (or any other value > 0)
Act
Hi,
I've noticed that a small but meaningful quota of calls from my Alcatel PBX to
Asterisk are failing.
This does not always happen and it is not easily reproducible but on high
traffic I do get a large number of cases.
Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 ove
The diagnostic tool is dahdi_test or -- if memory serves correctly
zttest. If you don't see a minimum of 99% after a minute or two, you
are in trouble.
hbk wrote:
> Thank you, very interesting!
>
> As I understand the Digium card is used as a interrupt source for Asterisk?
>
> Is there a d
Hi there,
i'm using dahdi to manage a B400P openvox BRI card.
All works as expected, i would like to know if there ia a way to put the
call in REMOTE HOLD, like pressing R button on ISDN phone.
This can be done by CAPI using the proper application ,
It is implemented on DAHDI ?
Regards Andrea
-
That's a pretty crappy phone huh? :)
Anyway you should be able to do it on features.conf, in the applicationmap
section. I'm not entirely sure there's a dialplan app that allows you to put a
channel on hold and take it back later.
Vinícius Fontes
www.asteriskforum.com.br - Informações e disc
look at Random()
2009/12/12 Landy Landy
> Hello List.
>
> I would like to know how I can use two or more service providers with
> asterisk to be used randomly for ei, if an user tries to make a call I would
> like to randomly use a provider. It doesn't matter where the call is
> destined to.
>
>
Hi everybody,
I have a sip phone (Siemens) which has no sip functions at all.
Is is possible to press #4 by example to put the call on hold then
dial #2 to get the call back ?
I'have look at features.conf but i did not find the solution.
I know the call parking functionnality, but i would lik
11 dec 2009 kl. 23.21 skrev John Taylor:
> I have multiple trunks to the same ITSP. Incoming calls to any trunk
> go to the last "incoming" label defined in those trunks' contexts in
> sip.conf.
>
> My ITSP insists on insecure=very in the trunk context; is this the cause?
>
This is an effect of
11 dec 2009 kl. 17.18 skrev Jerry Geis:
> Where in the code does something like:
> register => user[:secret[:authuse...@host[:port][/extension]
> from sip.conf 1) get parsed 2) actually register.
>
> I tried looking in channels/chan_sip.c and don't see where that happens.
> Ca
[Dec 14 09:23:18] ERROR[15198]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR
transmission error to XX.XX.XX.65:5037, rtcp halted Operation not
permitted
This is a log entry on a public Asterisk-server. My SIP-client
(Grandstream GXP2010) can perfectly register to this public
Asterisk-server.
My SIP-clie
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