[asterisk-users] SIp Signalling

[Il Neofita]

Is there a way to force asterisk to take care only of sip signaling without
forcing it to take care of rtp traffic?
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[asterisk-users] Realtime SIP

[Il Neofita]

Probably I did not read well the information
I am concerning, if I am going to use ARA for the SIP
and I have
register => user:secret:[EMAIL PROTECTED]:port/extension

how I should input that line?
If I am going to delete it from the DB I am forced to reload everything or
there is a way to tell asterisk to remove only a particular entry?
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[asterisk-users] Asterisk 1.4.15 Voicemail

[Il Neofita]

Hi
after having installed asterisk 1.4.15 my voicemail does not work anymore.
Am I the only one?
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Re: [asterisk-users] [Fwd: Re: VoiceMail hangup]

[Il Neofita]

Hi
I have the same problem

On Nov 13, 2007 9:10 AM, marcotasto <[EMAIL PROTECTED]> wrote:
> Hi Neofita, Doug and All.
>
> I think I've the same problem but I don't know if it's related to the bug 
> suggested below.
> I try to explain my behavior:
> - I dial the voicemail extension.
> - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... 
> or # to exit" (I listen the complete message or most part of it)
> - I press 1
> - I can hear the first recorded message.
>
> But, if:
> - I dial the voicemail extension.
> - I hear "you have 1 new message. Press 1..." 1 pressed (without waiting for 
> the message playing)
> - Asterisk hangups.
>
> I'm not always able to replicate the problem but, as "Il Neofita", I'm using 
> the italian prompts... could be a problem related to that?
>
> Bye and regards
>
> Marco Signorini.
>
>
>
>
> > Il Neofita wrote:
> > > --  Playing
> > > '/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language
> > > 'it')
> > >   == Spawn extension (servizi, , 1) exited non-zero on
> > > 'Local/[EMAIL PROTECTED],2'
> > >
> > >
> > It may be related to this bug:
> >
> > http://bugs.digium.com/view.php?id=11083
> >
> > Doug
> >
> >
> > --
> >
> > Ben Franklin quote:
> >
> > "Those who would give up Essential Liberty to purchase a little Temporary
> > Safety, deserve neither Liberty nor Safety."
> >
>
>
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Re: [asterisk-users] VoiceMail hangup

[Il Neofita]

Probably I was not able to explain myself properly
however, for some measge this what happen

--  Playing
'/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language
'it')
  == Spawn extension (servizi, , 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'

I cannot listen the message and the voicemailmain exists I am using
asterisk 1.4.13

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Re: [asterisk-users] VoiceMail hangup

[Il Neofita]

Thank you for your answer.
The problem is quite different
for example,
I am leaving a message of 5 seconds
when I call to listen the message , asterisk answer and pass the call
to voicemailmain and it plays the welcome message
now if I press 1 before 3 or 4 seconds the voicemailmain gives me then
information of the message
send the command to play the message and it exists.
If I wait more the 3 or 4 seconds and then I press 1 everything is
going well for the same kind of message

On Nov 12, 2007 3:53 PM, Anselm Martin Hoffmeister
<[EMAIL PROTECTED]> wrote:
> Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita:
>
> > Hi
> > additional information if I am going to wait at least 3 seconds after
> > the voicemail starts to give me the instruction I am able to listen my
> > messages.
> > But why I need to wait?
> >
> > On Nov 12, 2007 2:28 PM, Il Neofita <[EMAIL PROTECTED]> wrote:
> > > Hi,
> > > with some messages the voicemailmain after give me the information
> > > about the call (Days, hours and minutes) it finish.
> > >
> > > Whant can I check for solve this problem?
> > >
>
> Read voicemail.conf. Look for "minmessage" setting - it will remove
> messages that are shorter than the given number of seconds.
>
> See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
> See http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf
>
> BR
> Anselm
>
>
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Re: [asterisk-users] VoiceMail hangup

[Il Neofita]

Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?

On Nov 12, 2007 2:28 PM, Il Neofita <[EMAIL PROTECTED]> wrote:
> Hi,
> with some messages the voicemailmain after give me the information
> about the call (Days, hours and minutes) it finish.
>
> Whant can I check for solve this problem?
>

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[asterisk-users] VoiceMail hangup

[Il Neofita]

Hi,
with some messages the voicemailmain after give me the information
about the call (Days, hours and minutes) it finish.

Whant can I check for solve this problem?

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Re: [asterisk-users] Help with loop counting?

[Il Neofita]

Hi
I believe that
exten => s,7,GotoIf($[${trips}=4]?,8)

the , should be :

On 10/24/07, Phil Knighton <[EMAIL PROTECTED]> wrote:
>
>  Hi
>
> I have a situation where I want to be able to count how many times a
> caller goes round a loop of "Please hold...", "please continue to hold".  I
> have found an example on voip-info but I can't get it to work.  Not sure if
> I've got some syntax wrong somewhere?  All that happens at the moment, is I
> hit is the playback of "som-debug" at . Any ideas would be appreciated!
>
> extensions.conf:
>
> [so-mainmenu]
> exten => s,1,Answer
> exten => s,2,Set(trips=1)
> exten => s,3,SetMusicOnHold(default)
> exten => s,4,Set(TIMEOUT(digit)=5)
> exten => s,5,Set(TIMEOUT(response)=10)
> exten => s,6,Background(softopt/som-mainmenu)
> exten => s,7,GotoIf($[${trips}=4]?,8)
> exten => s,8,WaitExten(5)
> exten => s,9,Wait(5)
> exten => 1,1,Goto(so-sandm,s,1)
> exten => 2,1,Goto(so-support,s,1)
> exten => 3,1,Goto(so-accbill,s,1)
> exten => 4,1,Goto(so-switchboard,s,1)
> exten => 5,1,Goto(so-silentdial),s,1)
> exten => s,10,Background(softopt/som-mainmenuretry)
> exten => s,11,Wait(1)
> exten => s,12,Background(softopt/som-mainmenuopts)
> exten => s,13,Goto(s,7)
> exten => ,1,Playback(softopt/som-debug)
> exten => ,2,Hangup()
> exten => i,1,Set(trips=$[${trips} + 1])
> exten => i,2,Goto(s,7)
>
> Cheers
>
> Phil
>
> Phil Knighton
> Soft Option Technologies
>
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[asterisk-users] Force codec order

[Il Neofita]

There is a way to force the order of the codecs in the sip.conf since the
allow seams to let know only the accepted codec.
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[asterisk-users] Strange behaviour afetr update from 1.2 to 1.4

[Il Neofita]

Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a call
from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
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Re: [asterisk-users] GTALK problem

[Il Neofita]

Thank you
I need to wait the international version of gtalk


On 10/11/07, Philippe Sultan <[EMAIL PROTECTED]> wrote:
>
> > If I calling asterisk with GTALK in english everything is ok, however,
> some
> > of my friends with the italian version of gtalk they cannot have the
> audio.
>
> Audio problems might be experienced with older Gtalk clients. Version
> 1.0.0.104 is reported to work.
>
> The following resources may help you :
> http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Bugsampknownissues
> http://bugs.digium.com/view.php?id=10512
>
> Hope this will help you solve the problem,
>
> Philippe
>
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[asterisk-users] GTALK problem

[Il Neofita]

Hi,
I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from
1000 to 4
If I calling asterisk with GTALK in english everything is ok, however, some
of my friends with the italian version of gtalk they cannot have the audio.

Is it a bug? Or I did some mistake
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[asterisk-users] Curiosity Max Calls

[Il Neofita]

Hi
is there a tool to know what was the maximum calls that asterisk managed?

Thank you
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[asterisk-users] Call hangup after 60seconds

[Il Neofita]

Hi,
I have a client (xlite) connected to my server, on the server I have
type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the
server is listening on port 5060.
However, xlite is connect to a router where the port 5060 is blocked,
therefore, I am using 5065 and I have an iptables rule to transfer the
incoming packet from 5065 to 5060,
I cannot use the port 5065 since some ATA the do not allow the change of the
port.
When I am calling with xlite the call endup after 60seconds, but in the
60seconds I can talk.
Now if I am setting the client (in the sip.conf) in peer everything is
working.

Someone can explain to me why? What I am doing wrong?

Thank you
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Re: [asterisk-users] Strange Behaviour

[Il Neofita]

Thank you I will try tonight

On 9/10/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
>
> Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
> > On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]>
> > wrote:
> > Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
> >
> > Well, it seems there are differences between those accounts
> > then.
> >
> > You might want to post your sip.conf, and -if that is
> > possible- the ATA
> > conf file; or at least a writedown of the configuration there.
> >
> > First of all, thank you for you reply
> > The ATA is the Fritz!Box and I tried with different FW version but I
> > have the same behaviour
>
> I have been using FritzBoxes for quite a while, and have not found such
> strange bugs - except after a Firmware Upgrade. It seems after some
> upgrades you need to do a "factory reset" (via the web interface) and
> enter your data again, else they behave stupidly.
>
> > this is part of the sip.conf
> > [180]
> > type=peer
> > username=180
> > secret=aa
> > callerid=First<180>
> > canreinvite = yes
> > host = dynamic
> > dtmfmode = rfc2833
> > qualify = yes
> > nat = yes
> > context = mycont
> > disallow = all
> > allow = g726
> > allow = g723
> > allow = ulaw
> > allow = alaw
> > allow = g729
> > allow = gsm
> >
> > [181]
> > type=peer
> > username=181
> > secret=bb
> > callerid=Second<181>
> > canreinvite = yes
> > host = dynamic
> > dtmfmode = rfc2833
> > qualify = yes
> > nat = yes
> > context = mycont
> > disallow = all
> > allow = g726
> > allow = g723
> > allow = ulaw
> > allow = alaw
> > allow = g729
> > allow = gsm
>
> Looks pretty OK to me. Just a stupid idea: Do you have a [general]
> section before those two?
>
> And then, I use type=friend, not type=peer, that _might_ make a
> difference in how asterisk matches sip.conf contexts to registered
> clients.
>
> 8< From my sip.conf:
> [sip501]
> mailbox=01
> callerid=501
> type=friend
> username=sip501
> secret=lk1j2eu89
> context=sipclient
> host=dynamic
> nat=yes
> disallow=all
> allow=alaw
> allow=gsm
> allow=ulaw
>
> [sip502]
> mailbox=02
> callerid=502
> type=friend
> username=sip502
> secret=1092jd0
> context=sipclient
> host=dynamic
> nat=yes
> disallow=all
> allow=alaw
> allow=gsm
> allow=ulaw
> =>8
>
> Note: Those two accounts belong to the same FritzBox.
>
> > I tried to switch the account for the two ports but what it is
> > important is only the order in the sip.conf
>
> That made me think about that friend/peer thingy.
>
> > I found some information in german and I do not know it
>
> The FritzBoxes are popular here in Germany - no wonder, being a German
> manufactured product and being given away for (nearly) free with any
> 2-year DSL contract... I like them nevertheless :)
>
> BR, HTH
>
> Anselm
>
>
>
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Re: [asterisk-users] Strange Behaviour

[Il Neofita]

On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
>
> Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
>
> Well, it seems there are differences between those accounts then.
>
> You might want to post your sip.conf, and -if that is possible- the ATA
> conf file; or at least a writedown of the configuration there.


First of all, thank you for you reply
The ATA is the Fritz!Box and I tried with different FW version but I have
the same behaviour

this is part of the sip.conf
[180]
type=peer
username=180
secret=aa
callerid=First<180>
canreinvite = yes
host = dynamic
dtmfmode = rfc2833
qualify = yes
nat = yes
context = mycont
disallow = all
allow = g726
allow = g723
allow = ulaw
allow = alaw
allow = g729
allow = gsm

[181]
type=peer
username=181
secret=bb
callerid=Second<181>
canreinvite = yes
host = dynamic
dtmfmode = rfc2833
qualify = yes
nat = yes
context = mycont
disallow = all
allow = g726
allow = g723
allow = ulaw
allow = alaw
allow = g729
allow = gsm


If those are not the source of trouble, _I_ probably would switch the
> accounts in the ATA (port A versus port B) and try if the problem sticks
> with the port or with the account.


I tried to switch the account for the two ports but what it is important is
only the order in the sip.conf

I would also google if there are
> known problems with my ATA, look if a newer firmware is available, if
> there are informative messages that are worth a verbatim quote, and get
> another bottle of beer to keep the sunday relaxation at a proper level.


I found some information in german and I do not know it
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[asterisk-users] Strange Behaviour

[Il Neofita]

Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call only
with the second one in order to the sip.conf and the first it gives me 403.
And idea how to solve it?
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[asterisk-users] extra field

[Il Neofita]

Hi,
I am using my asterisk server like a gateway and one provider ask me to pass
an extra field with the IP of the peer that is using the connection,
probably to have more control on the authentication. I was wondering how I
can implement this.

Thank you
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Re: [asterisk-users] TDM400p Loaded only once

[Il Neofita]

Thank you for the answer
after

modprobe wctdm
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm

/proc/zaptel/ (empty)

/usr/src/asterisk/zaptel-1.2.14/xpp/utils/genzaptelconf -l (no result)



On 3/1/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:


On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
> Hi
> when I turn on my PC I able to load the drivers and start my card,
> if I reboot the PC I have the following error
>
> ztcfg -vvv
>
> Zaptel Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
> Channel 03: FXS Kewlstart (Default) (Slaves: 03)
> Channel 04: FXS Kewlstart (Default) (Slaves: 04)

This is what you attempt to configure

>
> 4 channels configured.
>
> ZT_CHANCONFIG failed on channel 1: No such device or address (6)

And that is the result.

What do you see on /proc/zaptel ?

What is the output of 'xpp/utils/genzaptelconf -l' from the zaptel
source directory?

>
> This is part of my dmesg
> audit(1172747900.510:5): avc:  denied  { net_bind_service }
for  pid=1657
> comm="hidd" capability=10 scontext=system_u:system_r:bluetooth_t:s0
> tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
> SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
> eth0: no IPv6 routers present
> [drm] Initialized drm 1.1.0 20060810
> ACPI: PCI Interrupt :01:00.0[A] -> GSI 16 (level, low) -> IRQ 19
> [drm] Initialized r128 2.5.0 20030725 on minor 0
> agpgart: Found an AGP 3.0 compliant device at :00:00.0.
> agpgart: Device is in legacy mode, falling back to 2.x
> agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
> agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
> audit(1172747921.184:6): avc:  denied  { getattr } for  pid=2323
> comm="pam_console_app" name="card0" dev=tmpfs ino=7969
> scontext=system_u:system_r:pam_console_t:s0-s0:
c0.c255tcontext=system_u:object_r:device_t:s0
> tclass=chr_file
> Zapata Telephony Interface Registered on major 196
> Zaptel Version: 1.2.14
> Zaptel Echo Canceller: KB1

Load of zaptel. No load of wctdm in sight.

>
> and finally this is my configuration
> fxsks=1-4
> loadzone=us
> defaultzone=us

--
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+972-50-7952406   mailto:[EMAIL PROTECTED]
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[asterisk-users] TDM400p Loaded only once

[Il Neofita]

Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error

ztcfg -vvv

Zaptel Configuration
==


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

This is part of my dmesg
audit(1172747900.510:5): avc:  denied  { net_bind_service } for  pid=1657
comm="hidd" capability=10 scontext=system_u:system_r:bluetooth_t:s0
tcontext=system_u:system_r:bluetooth_t:s0 tclass=capability
SELinux: initialized (dev autofs, type autofs), uses genfs_contexts
eth0: no IPv6 routers present
[drm] Initialized drm 1.1.0 20060810
ACPI: PCI Interrupt :01:00.0[A] -> GSI 16 (level, low) -> IRQ 19
[drm] Initialized r128 2.5.0 20030725 on minor 0
agpgart: Found an AGP 3.0 compliant device at :00:00.0.
agpgart: Device is in legacy mode, falling back to 2.x
agpgart: Putting AGP V2 device at :00:00.0 into 1x mode
agpgart: Putting AGP V2 device at :01:00.0 into 1x mode
audit(1172747921.184:6): avc:  denied  { getattr } for  pid=2323
comm="pam_console_app" name="card0" dev=tmpfs ino=7969
scontext=system_u:system_r:pam_console_t:s0-s0:c0.c255tcontext=system_u:object_r:device_t:s0
tclass=chr_file
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.14
Zaptel Echo Canceller: KB1

and finally this is my configuration
fxsks=1-4
loadzone=us
defaultzone=us
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[asterisk-users] Strange Noise

[Il Neofita]

Hi,
today with my asterisk during a call I had a very strange noise, it was the
typical noise that you can have  when your device uses a bad power supply.
I change phone and I had the same behavior after I while I tried again and
the noise was disappeared.

What can I check?
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Re: [asterisk-users] Strange behaviour with Dial cmd

[Il Neofita]

Yes, but I would like to try a number and after to try a second one.
Any Idea how to avoid this.

On 2/18/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:


C F wrote:
> Asterisk supports this directly by issuing the hangup command before
> the answer command.  However, when using an analog interface like FXO
> the line has no way of knowing you just hung up and will continue to
> ring, which asterisk will see as a new call. in my experience even
> when using a PRI if i dont give the pri cause the provider re
> initiates the call.

This would only happen if you blindly run two Dial lines in sequence in
your dialplan.  Don't do this.
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Re: [asterisk-users] Strange behaviour with Dial cmd

[Il Neofita]

Ok thank you a lot!!!

On 2/15/07, Yuan LIU <[EMAIL PROTECTED]> wrote:


>From: "Il Neofita" <[EMAIL PROTECTED]>
>Date: Thu, 15 Feb 2007 03:37:14 -0500
>
>But I tought that hangup was suppose to close the call, however, is not
the
>case and a really did not catch why.

Now I see where the confusion comes from.  Asterisk doesn't really speak
English - or Chinese for that matter:-)  In telephony, there is no way for
the callee to tell the caller to stop ringing - unless you "answer" it
first.  Once you answer, you can do a number of things, the rudest being
to
immediately hang up. (I saw live people doing this intentionally.)  Your
only other option really is to ignore.

I just thought up this simple method to ignore: divert the dial plan to
simply Wait() an unreasonable amount of time in hope that the caller hangs
up.

exten => s,1,Dial(yourcell,5)
exten => s,n,Wait(300)

That's assuming your provider provides disconnect supervision.  You can
also
Play(prerecorded,noanswer) if your provider supports it. (Won't hurt to
try
even if not - but disconnect supervision is a must.)

Yuan Liu


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Re: [asterisk-users] Strange behaviour with Dial cmd

[Il Neofita]

On 2/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote:



Well, you'll have to decide how you want to "hang up" the caller: Do you
want him/her to be ignored, or to be told that you are not available (like
an answering machine)?  You also need to tell  Asterisk how to determine
if
the next "invite" comes from the same caller during the same "session".
(These are not very easy tasks but doable.)

In either case, you need to add a flag to your dial plan, set it after it
rings your cell, and reset it once Asterisk determines that THIS caller
has
been "hung up". (Of course you can do what "Vacation" does in E-mail: set
up
a flag for each identifiable caller, and only call your mobile once until
you reset them all.  The algorithm would be simpler but more
unidentifiable
callers will be ignored.)  Your dial plan will check this flag before
ringing your cell, then branch accordingly.

Hope this helps.

Yuan Liu



Thank you for the answer.
Right know I solved sending after everything to the voicemail.
But I tought that hangup was suppose to close the call, however, is not the
case and a really did not catch why.

I will try to read a bit more.
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Re: [asterisk-users] Strange behaviour with Dial cmd

[Il Neofita]

This is the situation
A call me at my provider 1
I am not home and I would like to transfer the call
I do not pickup the call for some reason
I would like to hangup the caller, however, my asterisk try again to call on
my mobile over and over

I would like to stop it.

Any idea?

Thank you a lot.

On 2/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote:


>From: "Il Neofita" <[EMAIL PROTECTED]>
>Date: Wed, 14 Feb 2007 19:30:51 -0500
>
>I have this simple context
>
>I am register to an external provider and when I am not home I would like
>to transfer the phone outside
>The problem that the call goes in loop

I don't see any loop in records below?  What your dial plan does is what
you
told it to: ring for 5 sec, then hang up.  Because the calling party
hasn't
given up, your dial plan rings again.  Where's the error?

Yuan Liu

>I cannot understand why.
>
>Can you figure out my error?
>
>Thank you
>
>sip.conf
>register => user:[EMAIL PROTECTED]/400
>
>[inside]
>exten => _4X.,1,dial(SIP/ext_400_124/555123,5,tT)
>exten => _4X.,2,hangup
>
>-- Executing Dial("SIP/555123-081b8eb0",
>"SIP/ext_400_124/555123|5|tT") in new stack
>-- Called ext_400_124/555123
>-- SIP/ext_400_124-081bf848 is ringing
>-- Nobody picked up in 5000 ms
>-- Executing Hangup("SIP/555123-081b8eb0", "") in new stack
>  == Spawn extension (inside, 5123, 2) exited non-zero on
>'SIP/555123-081b8eb0'
>-- Executing Dial("SIP/555123-081c79a8",
>"SIP/ext_400_124/555123|5|tT") in new stack
>-- Called ext_400_124/555123
>-- SIP/ext_400_124-081ccee8 is ringing
>  == Spawn extension (inside, 5123, 1) exited non-zero on
>'SIP/555123-081c79a8'


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[asterisk-users] Strange behaviour with Dial cmd

[Il Neofita]

I have this simple context

I am register to an external provider and when I am not home I would like to
transfer the phone outside
The problem that the call goes in loop

I cannot understand why.

Can you figure out my error?

Thank you


sip.conf
register => user:[EMAIL PROTECTED]/400

[inside]
exten => _4X.,1,dial(SIP/ext_400_124/555123,5,tT)
exten => _4X.,2,hangup

-- Executing Dial("SIP/555123-081b8eb0",
"SIP/ext_400_124/555123|5|tT") in new stack
   -- Called ext_400_124/555123
   -- SIP/ext_400_124-081bf848 is ringing
   -- Nobody picked up in 5000 ms
   -- Executing Hangup("SIP/555123-081b8eb0", "") in new stack
 == Spawn extension (inside, 5123, 2) exited non-zero on
'SIP/555123-081b8eb0'
   -- Executing Dial("SIP/555123-081c79a8",
"SIP/ext_400_124/555123|5|tT") in new stack
   -- Called ext_400_124/555123
   -- SIP/ext_400_124-081ccee8 is ringing
 == Spawn extension (inside, 5123, 1) exited non-zero on
'SIP/555123-081c79a8'
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Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk

[Il Neofita]

Sei riuscito?
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Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk

[Il Neofita]

On 2/10/07, Patrick <[EMAIL PROTECTED]> wrote:


Where can I find that patch?

Thanks,
Patrick



Hi Patrick,
I downloaded the patch from here

http://bugs.digium.com/view.php?id=7764
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Re: [asterisk-users] RE: asterisk 1.4 FC5 and Gtalk

[Il Neofita]

Hi,
I tried thousands of time and finally I am a step closer to the solution.
I recompile iksemel with the option --prefix=/usr
I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4, re-extracting
everything from the tar
recompile everything and now jabber is working or almost. When I received a
call asterisk crash, but I saw that there is a patch and I am applying that
right know.

See you

On 2/9/07, Mani Sridhar <[EMAIL PROTECTED]> wrote:


i saw the same problem and here is a thread where i mentioned how i fixed
it..

http://lists.digium.com/pipermail/asterisk-users/2006-November/171783.html

look for my previous mails in this thread sometime september-november 2006
.

btw, i can't get asterisk to work with google talk yet.

thanks
sridhar

>From: [EMAIL PROTECTED]
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: asterisk-users Digest, Vol 31, Issue 39
>Date: Fri,  9 Feb 2007 18:30:38 -0700 (MST)
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>
>Send asterisk-users mailing list submissions to
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>
>To subscribe or unsubscribe via the World Wide Web, visit
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>or, via email, send a message with subject or body 'help' to
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>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of asterisk-users digest..."
>
>
>
>Message: 12
>Date: Fri,  9 Feb 2007 23:28:15 +0100
>From: "marcotasto" <[EMAIL PROTECTED]>
>Subject: Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk
>To: "asterisk-users" 
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset=iso-8859-1
>
>Ciao Neofita.
>I'm trying my GTalk account and I'm still having the same problem.
>I've installed the gnuTLS-developer rpms and rebuilt and re-installed the
>complete Asterisk package but without success.
>I'm working with OpenSuse 10.2.
>
>This is my debug info that's quite similar to what you've posted:
>
>JABBER: asterisk OUTGOING: xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client'
>to='gmail.com' version='1.0'>
>Gateway*CLI>
>JABBER: asterisk INCOMING: encoding="UTF-8"?>version="1.0" xmlns:stream="http://etherx.jabber.org/streams";
>xmlns="jabber:client">xmlns="urn:ietf:params:xml:ns:xmpp-tls"/>xmlns="urn:ietf:params:xml:ns:xmpp-sasl">X-GOOGLE-TOKEN
>[Feb  9 23:15:43] ERROR[24214]: res_jabber.c:482 aji_act_hook: gnuTLS not
>installed.
>
>There is someone knowing what's the problem and that could help us?
>
>Best regards,
>
>Marco Signorini.
>
>
>
>
>--
>Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom
>http://click.libero.it/infostrada9feb07
>
>
>
>

_
Gossips, movie reviews, photogallery and more
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[asterisk-users] asterisk 1.4 FC5 and Gtalk

[Il Neofita]

JABBER: gtalk_account OUTGOING: 
localhost*CLI> jabber show tes
JABBER: gtalk_account INCOMING: http://etherx.jabber.org/streams";
xmlns="jabber:client">
[Feb  9 21:11:15] ERROR[2061]: res_jabber.c:482 aji_act_hook: gnuTLS not
installed.

I installed all the gnutls but I still have this error


[EMAIL PROTECTED] ~]# rpm -qa | grep gnutls
gnutls-utils-1.2.10-3
gnutls-1.2.10-3
gnutls-devel-1.2.10-3


Do you know how to solve it?
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[asterisk-users] Re: Chan_Cellphone

[Il Neofita]

I start the patch and automatically created the file. But now on the menu I
cannot select chan_cellphone
I launched ./bootstrap.sh
and after ./configure
in my /usr/include/bluetooth I have the header
but I cannot select the option

any idea?

On 2/9/07, Il Neofita <[EMAIL PROTECTED]> wrote:


Hi,
I download the last svn and I also look around but I cannot find the
source, I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919

any one can help me out.

thx

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Re: [asterisk-users] BindPort

[Il Neofita]

The point is to use more than one port, I think the only way is to use the
redirect from iptables

On 2/6/07, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:


Ciao,
just change port value in sip.conf.

Giorgio

Il Neofita wrote:
> Hi,
> I was wondering if it is possible to set asterisk in order to listen
> to different ports for the sip or I need to do this operation with
> iptables?
> All of this since some time the port 5060 is blocked.
>
> Thank you
> 
>
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[asterisk-users] Chan_Cellphone

[Il Neofita]

Hi,
I download the last svn and I also look around but I cannot find the source,
I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919

any one can help me out.

thx
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[asterisk-users] BindPort

[Il Neofita]

Hi,
I was wondering if it is possible to set asterisk in order to listen to
different ports for the sip or I need to do this operation with iptables?
All of this since some time the port 5060 is blocked.

Thank you
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Re: [asterisk-users] chanskype

[Il Neofita]

Hi,
I was wondering if someone had  problems with chanskype.
Since I am wondering if they are a credible company or not.

See you
On 1/20/07, Moises Silva <[EMAIL PROTECTED]> wrote:


> Hi,
> I tried the try version of chanskype, however, everytime that I make a
call
> asterisk generate an error
So you think is easy to us guess wich error you are getting?
Seriously, I think you should read this:
http://www.catb.org/~esr/faqs/smart-questions.html

> Anyone has experience with this? Since I tried to contact the support
but
> they never replied.

Please provide more information about the error, and search in
voip-info.org how to raise the verbosity level of asterisk in the
console.

Also sometimes help searching the error in google.

Regards


--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org";
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[asterisk-users] chanskype

[Il Neofita]

Hi,
I tried the try version of chanskype, however, everytime that I make a call
asterisk generate an error

Anyone has experience with this? Since I tried to contact the support but
they never replied.

Thank you
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Re: [asterisk-users] Looking for toll free in Italy

[Il Neofita]

I do not think that there are some company that offer a toll free number
(Numero verde in italian)
But contact on of these three providers
http://www.eutelia.it/tlc/
http://www.unidata.it/
http://messagenet.it/

If they have one of these should be able to give to you

See you

On 1/8/07, CM Rahman <[EMAIL PROTECTED]> wrote:




*Hi,*
**
*I am looking for tollfree number in italy. Anybody providing that? Charge
per minute? It will connect to my asterisk pbx box.*
**
*Thanks*
**
*CM*

__
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http://mail.yahoo.com

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[asterisk-users] Strange error

[Il Neofita]

Someone know why my asterisk gives me the following msgs?
Thank you

- Got SIP response 603 "Declined (no dialog)" back from
X.X.X.X
   -- Got SIP response 603 "Declined (no dialog)" back from
X.X.X.Y
   -- Got SIP response 603 "Declined (no dialog)" back from
X.X.X.Z
   -- Got SIP response 603 "Declined (no dialog)" back from
X.X.X.X
   -- Got SIP response 603 "Declined (no dialog)" back from
X.X.X.X
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Re: [Asterisk-Users] WebPhone

[Il Neofita]

On 7/3/06, Jean-Denis Girard <[EMAIL PROTECTED]> wrote:
But I'm not sure that MozPhone is what the original poster asked.No, however, I like to read all these different point of view.
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[Asterisk-Users] WebPhone

[Il Neofita]

Hi,someone know a good webphone, possibily a free oneThx
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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer <[EMAIL PROTECTED]
> wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd  on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.
pFrom: "Il Neofita" <
[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
asterisk-users@lists.digium.com>Date: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30)    
exten =>
 _40002,1,Dial(SIP/40002,30)     From: "Il Neofita" <[EMAIL PROTECTED]
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com
>Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console    -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack
    -- Called 40002    -- SIP/40002-4753 is ringing    -- SIP/40002-4753 answered SIP/40001-3760     -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI> sip show channels
Peer User/ANR    Call ID  Seq (Tx/Rx)  Form  Hold Last Message82.X2.XX3.X3
 40002   146b518a4cd  00103/0  alaw  No   Tx: ACK 82.X2.XX3.X3 40001   CBD1DB85-8B  00102/30987  alaw  No   Tx: ACK2 active SIP channels
 
		Do you Yahoo!? Everyone is raving about the 
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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

cosa vedo a console    -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack    -- Called 40002    -- SIP/40002-4753 is ringing    -- SIP/40002-4753 answered SIP/40001-3760
    -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI> sip show channelsPeer User/ANR    Call ID  Seq (Tx/Rx)  Form  Hold Last Message82.X2.XX3.X3 40002   146b518a4cd  00103/0  alaw  No   Tx: ACK
82.X2.XX3.X3 40001   CBD1DB85-8B  00102/30987  alaw  No   Tx: ACK2 active SIP channels
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Re: [Asterisk-Users] Canreinvite

[Il Neofita]

This is the dial in extensionsexten => _40001,1,Dial(SIP/40001,30)    exten => _40002,1,Dial(SIP/40002,30)    
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[Asterisk-Users] Canreinvite

[Il Neofita]

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come?
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[Asterisk-Users] Zaptel Module.symvers missing

[Il Neofita]

Hi,when I am going to compile the zaptel I receive this messageake -C /lib/modules/2.6.12-12mdk/build SUBDIRS=/usr/src/asterisk/zaptel-1.2.5 XPPMOD= modulesmake[1]: Entering directory `/usr/src/linux-2.6.12-12mdk
'  WARNING: Symbol version dump /usr/src/linux-2.6.12-12mdk/Module.symvers   is missing; modules will have no dependencies and modversions...lrwxrwxrwx   1 root  root    18 mag 23 17:42 linux -> 
linux-2.6.12-12mdk/drwxr-xr-x  22 root  root  4096 mag 23 18:04 linux-2.6.12-12mdk/..Any idea how to generate this file?cat /proc/versionLinux version 2.6.12-12mdk (
[EMAIL PROTECTED]) (gcc version 4.0.1 (4.0.1-5mdk for Mandriva Linux release 2006.0)) #1 Fri Sep 9 18:15:22 CEST 2005
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Re: [Asterisk-Users] NOTIFY Problem

[Il Neofita]

I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom <
[EMAIL PROTECTED]> wrote:Il Neofita wrote:> Hi,> one of my WiFI phone has problem with the notify asterisk signal to me
> the following> Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '*MailScanner warning:> numerical links are often malicious:* 192.168.100.124> <
http://192.168.100.124>' does not implement 'NOTIFY'>> In theory the phone support this function.>> Any idea?>If you remove the mailbox= bit of sip.conf for that host, then asterisk
will almost never send it a notify.Either way, is it causing a problem?___--Bandwidth and Colocation provided by Easynews.com
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[Asterisk-Users] NOTIFY Problem

[Il Neofita]

Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY'
In theory the phone support this function.Any idea?
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[Asterisk-Users] Remote UNIX connection disconnected over and over

[Il Neofita]

Hi,I am pretty sure that you already answer to this question, but I was not able to find the solutionon the console I have over and over the following msgs -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected
    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected
    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected    -- Remote UNIX connection disconnected
Any idea?
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[Asterisk-Users] How to use the cmd SMS

[Il Neofita]

Hi,I try to use my phone that has a SMS capability with asterisk.I am not able to receive SMS, someone can help me out?This what I am able to have but nothing more -- Executing SMS("SIP/503-7d2e", "508|sa") in new stack
    -- SMS TX 93 00 6D
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[Asterisk-Users] chan_gsm_bt Impression

[Il Neofita]

Hi,Has anyone proved the chan_gsm_bt ??Any impression?
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[Asterisk-Users] Error Header field Via

[Il Neofita]

Someone know the meaing of this error? chan_sip.c:3853 copy_via_headers: No header field 'Via' present to copy
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[Asterisk-Users] Codec Problem

[Il Neofita]

I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this error Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)
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Re: [Asterisk-Users] H323 on way voice

[Il Neofita]

This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI> h.323 debugH323 debug enabledmygw== New H.323 Connection created.    -- Setting up Call    -- oCall token:  [ip$192.168.1.219:1057/8226]
    -- oCalling party name:  [myPersonal]    -- oCalling party number:  []    -- oCalled party name:  [ip$192.168.1.214:1720]    -- oCalled party number:  []mygw--Received SETUP messageAllowed Codecs:
mygw Table:   G.729A <1>   G.729 <2>>   G.723.1 <3>   G.711-uLaw-64k <4>   G.711-ALaw-64k <5>   UserInput/hookflash <6>   UserInput/RFC2833 <7>
 Set:aco*CLI>   0:aco*CLI> 0:o*CLI>   G.729A <1>   G.729 <2>   G.723.1 <3>   G.711-uLaw-64k <4>   G.711-ALaw-64k <5> 1:o*CLI>
   UserInput/hookflash <6> 2:o*CLI>   UserInput/RFC2833 <7>mygw*CLI>mygw=-= In OnAnswerCall for call 8226mygw*CLI>   - Progress Indicator: 0mygw*CLI>   - Inserting PI of 0 into ALERTING message
  == Starting H323/ip$192.168.1.219:1057/8226 at default,ip$192.168.1.214:1720,1 failed so falling back to exten 's'    -- Executing Playback("H323/ip$192.168.1.219:1057/8226", "demo-echotest") in new stack
mygwAnswering call ip$192.168.1.219:1057/8226    -- Playing 'demo-echotest' (language 'en')mygw-- Started logical channel: sending G.711-uLaw-64kmygw*CLI>   -- channelsOpen = 1mygw=-= In OnConnectionEstablished for call 8226
mygw*CLI>   -- Connection Established with "myPersonal [192.168.1.219]"mygwMyH323_ExternalRTPChannel::OnReceivedAckPDUmygw*CLI>   -- remoteIpAddress: 
192.168.1.219    -- remotePort: 49600    External RTP Session Starting    RTP channel id 1 parameters:    -- remoteIpAddress: 
192.168.1.219    -- remotePort: 49600    -- ExternalIpAddress: 192.168.1.214    -- ExternalPort: 17950mygw*CLI> .    -- Executing Echo("H323/ip$192.168.1.219:1057/8226", "") in new stack

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[Asterisk-Users] H323 on way voice

[Il Neofita]

Hi,I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP.
Any thoughts?
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[Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!

[Il Neofita]

I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
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[Asterisk-Users] H323 behind a Firewall

[Il Neofita]

There is a proble to put an H323 Asterisk server behind an iptables firewall?
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[Asterisk-Users] H323 Info

[Il Neofita]

Hi,I compiled for my asterisk 1.2.4 the openh323but when I give this commandh.323 show codecsI receive thisAllowed Codecs: Table: Set:I cannot test with msn if everything is working since I am outside and I cannot access to the firewall.
Someone can tell me if I need to install the oh323 since I do not know if I need it or notThank you
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Re: [Asterisk-Users] TDM400P busy

[Il Neofita]

Yes, I changed.Thank youOn 3/27/06, Infobox Peru <[EMAIL PROTECTED]> wrote:
Maybe your lines use polarity reversals for hangup detection.On 3/27/06, 
Il Neofita <[EMAIL PROTECTED]
> wrote:
Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied.


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[Asterisk-Users] TDM400P busy

[Il Neofita]

Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied.
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[Asterisk-Users] AstCC

[Il Neofita]

Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you
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[Asterisk-Users] MWI problem

[Il Neofita]

Sorry if I am always here asking for MWI, but I do not know how to
solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they
think that I have a message waiting.
Anyone knows how to solve this issue?

Thank you
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[Asterisk-Users] WMI Problem

[Il Neofita]

Hi,
I was not able to find any indication for this problem that I have right now.
My phones connected to an Azatel 200 they always indicate that I have a message waiting to be listen.
However, I do not have any message.
I also checked using the console "show voicemail user for context" but I have 0 messages.
Any idea what I need to check?

I am using Asterisk 1.2.1

Thank you
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[Asterisk-Users] ATA does not register

[Il Neofita]

I am not able to register an  external ATA on my asterisk 2.0 Beta

This is the debug
Any idea?

server01*CLI>
<-- SIP read from CLIENTIP:5060:
REGISTER sip:SIPSERVERIP SIP/2.0
Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2
From: ;tag=1564789518
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon ata 11.03.37
Supported: 100rel, replaces
Allow-Events: telephone-event
Allow-Events: refer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER
Accept: application/sdp
Accept-Encoding: identity
Content-Length: 0

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[Asterisk-Users] WMI problem

[Il Neofita]

I installed astersik 1.2beta and from that point the led that indicate a new call flash.
The ATA installed is an AZATEL.
Any idea what can I check?
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[Asterisk-Users] TXFAX

[Il Neofita]

Hi,
do you know what it means the following:

 "Call failed to go through, reason 3"


I received it when I try to send a FAX and no one answer it.

Thank you
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Re: [Asterisk-Users] Txfax

[Il Neofita]

This is the script I do not remeber the name of the author

For me is working
On 9/9/05, Chris Shipman <[EMAIL PROTECTED]> wrote:
Thanks Luki,Maybe it was the wrong place to mention it.Back to the other matter: The program will create a PDF or tif. Itcan submit the fax by FTP or by AstFax.If someone is willing to develop a CGI/PERL script for HTTP submissions I
will integrate that into the program as well.Since my program uses a lot of Open source programs I will release itunder the GPL as well.  I plan to test the faxing portionsnext week.After I have some of the bugs worked out I will release the
program.Regards,Chris- Original Message -From: "Luki" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, September 09, 2005 6:20 PMSubject: Re: [Asterisk-Users] Txfax> > I don't understand, what you are telling or asking us
> > with this information. Has it something to do with your question?> But others might understand... I do. Yes, it's related.>> Essentially it's the equivalent of WHFC for Hylafax -- a Windows
> client that will allow Windows users print from any program to a> virtual fax printer; Chris's program will then submit the job as a> TIFF to the asterisk server along with a call file. Here's the
> relevant part: it's quite important to know if the fax went through or> not, and if it should be rescheduled for resending without (Windows)> user intervention.>> Anyway, IMO Roger's hack is useful regardless and I'd be interested in
> trying Chris's solution out once it's ready (and if he chooses to> share it).>> --Luki> ___> --Bandwidth and Colocation sponsored by 
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asterfax.0.22.tar.gz
Description: GNU Zip compressed data
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Re: [Asterisk-Users] sending fax

[Il Neofita]

HI Chris
I am interested, I would like to know how I can have the opportunity to test your program.
On 9/9/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote:> I started working on a program using Ghostscript and Redmon to generate> the tif in windows by a printer.> So far I am using FTP to transfer the tiff and call file.  At least until I
> figure something better out.Why don't you look at IPP (Internet Printing Protocol)? a protocol forsubmitting jobs over HTTP of some sort. Server is already implemented ine.g. cups.HTTP allows a nice header with some extra fields. I wonder if that can
be abused to get the call information through. (and am I re-inventingsome wheels in the process?)--Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il
|  
| a Mutt's[EMAIL PROTECTED]
|  
|  bestICQ#
16849755
|  
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Re: [Asterisk-Users] sending fax

[Il Neofita]

The site is in italian, and you need to register your self in order to download the script.
The script is in perl and you need to start in order to simulate the hylafax daemon after that you can use WHFC
On 9/8/05, Matthew Gibson <[EMAIL PROTECTED]> wrote:
Sorry to interrupt :)But I believe what you guys are searching for lays here:http://www.inter7.com/?page=astfaxThanks,MattWiley Siler wrote:
> Google can translate if that helps...>> w>>___--Bandwidth and Colocation sponsored by Easynews.com
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[Asterisk-Users] Txfax

[Il Neofita]

Is there some way to know if the fax was received correctly or not?
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[Asterisk-Users] Extension a

[Il Neofita]

Hi,
I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code.
Where I need to put this pass?
I am using asterisk 1.2.0 beta 1

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Re: [Asterisk-Users] Asterisk architecture

[Il Neofita]

The call generate from branch2 can be send to the asterisk in Branch1
with a trunk the same think the call received from branch1 the only
thing that is not cleat  how you want transfer automatically the
call received from the pstn. What rule you want use?On 9/5/05, housi mueller <[EMAIL PROTECTED]> wrote:
Why not? How would you solve then the Brench1/Branch2 issue??
a <[EMAIL PROTECTED]> wrote:
>From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller <[EMAIL PROTECTED]
> wrote:


I am new with asterisk and hope somebody can help me.
 
Is a configuration like shown on the picture with asterisk
correct?  Some phone calls arriving in Branch 1 should be
redirected automatically  to Branch 2 and all phone calls made
from Branch 2 should going out  over Branch 1. (Branch 2 is not
connected directly with a PSTN.)
 
Thank you in advace
 
Housi Mueller
 


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Re: [Asterisk-Users] Asterisk architecture

[Il Neofita]

>From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller <[EMAIL PROTECTED]> wrote:

I am new with asterisk and hope somebody can help me.
 
Is a configuration like shown on the picture with asterisk
correct?  Some phone calls arriving in Branch 1 should be
redirected automatically  to Branch 2 and all phone calls made
from Branch 2 should going out  over Branch 1. (Branch 2 is not
connected directly with a PSTN.)
 
Thank you in advace
 
Housi Mueller
 
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Re: [Asterisk-Users] sending fax

[Il Neofita]

Hi,
I found on a forum a script that "emulate a hylafax" this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423
You can use the WHFC in order to send a fax to asterisk.

On 9/5/05, Harald Klein <[EMAIL PROTECTED]> wrote:
Hi Chris, Hi Arne,Am 5.9.2005 schrieb "Chris Shipman" <[EMAIL PROTECTED]>:>I've  seen some programs that install as a printer and create an image.
>However this would be to cumbersome for your average user.>It would need to be able to print to as local printer and then send out>Asterisk.What about:Client with Postscript printer driver
Some kind of a printing system (samba with lpr[ng] and/or cups etc.) toaccess the fax-printer via smb/cifs/lpr/ipp/whatever..Output filter for the fax-printer to convert Postscript to tiff andgeneratea call file with App txfax...
The problem is to tell the printer the number to fax to...You can grep in the Postscript file for a predefined string (for example"Fax Recpient Nr") and generate some matching templates in your office
suite..Search for HylaFax solutions, they are pretty much the same...Hari>>Chris>>- Original Message ->From: "Arne Morten Johansen" <
[EMAIL PROTECTED]>>To: "Asterisk Users Mailing List - Non-Commercial Discussion">>Sent: Monday, September 05, 2005 6:27 AM
>Subject: SV: [Asterisk-Users] sending fax What about faxing yourself if you don't have a scanner? -Opprinnelig melding->> Fra: 
[EMAIL PROTECTED]>[mailto:[EMAIL PROTECTED]] På vegne av Johan van>Tongeren>> Sendt: 5. september 2005 09:11
>> Til: Asterisk Users Mailing List - Non-Commercial Discussion>> Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing]>> exten => s,1,SetCIDNum(0${CALLERIDNUM})
>> exten => s,2,Dial(Zap/g${ARG2}/${ARG1},20,,t)>> exten => s,3,Goto(900)>> exten => s,103,Goto(900)>> exten => s,900,Busy>> exten => s,901,Hangup>>
>> -Oorspronkelijk bericht->> Van: [EMAIL PROTECTED]>> [mailto:
[EMAIL PROTECTED]] Namens Chris Shipman>> Verzonden: maandag 5 september 2005 7:22>> Aan: Asterisk Users Mailing List - Non-Commercial Discussion>> Onderwerp: [Asterisk-Users] sending fax
 I've read alot on the wiki about sending and receiving faxes thru>> asterisk.>> I've gotten the receive to work great.My question is how does one>> send a>> fax?
>> I see lots of instructions about how to send the image to asterisk by>> email,>> etc.  The problem is how does  one make the image of the fax to>> begin>> with?   Has anyone come up with a good solution for this?
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Re: [Asterisk-Users] Problem with include

[Il Neofita]

Sorry, I'm using the 1.0.9On 9/1/05, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Il Neofita wrote:> Hi,> I put on sip.conf the following line>> #include "sip.d/*.conf"You neglected to include the most important piece of information: whatversion of Asterisk you are using.
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[Asterisk-Users] Problem with include

[Il Neofita]

Hi,
I put on sip.conf the following line

#include "sip.d/*.conf"

inside I have files like that

provider1.conf
provider2.conf

But asterisk does not want to load it
This is the error

Sep  1 13:18:35 VERBOSE[8756]:   == Parsing
'/etc/asterisk/sip.d/*.conf': Sep  1 13:18:35
VERBOSE[8756]:   == Parsing '/etc/asterisk/sip.d/*.conf': Not
found (No such file or directory)

this is the ls result

[EMAIL PROTECTED] asterisk]# ls /etc/asterisk/sip.d/ -la
total 13
drwxrwxrwx  2 asterisk asterisk 4096 Sep  1 13:06 ./
drwxr-xr-x  9 asterisk asterisk 4096 Sep  1 13:17 ../
-rwxrwxrwx  1 asterisk asterisk  276 Sep  1 13:06 provider1.conf*
-rwxrwxrwx  1 asterisk asterisk  274 Sep  1 13:06 provider2.conf*


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[Asterisk-Users] TXFAX() status

[Il Neofita]

Hi,
I'm using  a script in order to send out my faxes with the application
txfax, therefore, I do not know how to see if the faxes are sent.
Any idea?
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Re: [Asterisk-Users] ASTCC and cdrs

[Il Neofita]

Thank you was that the problem.

On 8/25/05, Darren Wiebe <[EMAIL PROTECTED]> wrote:
> When did you install it?  Try running the "update database" function
> from the configure menu.
> 
> Darren Wiebe
> [EMAIL PROTECTED]
> 
> Il Neofita wrote:
> 
> >My installation of ASTCC does not update the cdrs tables .
> >It is a problem of ASTCC or it is a configuration problem?
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[Asterisk-Users] ASTCC and cdrs

[Il Neofita]

My installation of ASTCC does not update the cdrs tables .
It is a problem of ASTCC or it is a configuration problem?
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[Asterisk-Users] Calling Card Application

[Il Neofita]

I am looking for a calling card application which is able to advise me
during a call when the credit is almost finish. For examples 1 minute
before the end of the credit.

Thank you
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[Asterisk-Users] X100P compatible

[Il Neofita]

Why my X100P detect the ring after 3 o 4 seconds?
The funny thing that when I have an incoming call asterisk receive a
signal but the commands start after 3 or 4 seconds. Moreover, when the
call end the hungup has the same delay.
any ideas?
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Re: [Asterisk-Users] initiating Monitor during call

[Il Neofita]

I put these lines on features.conf in asterisk CVS-v1-0-08/16/05
[featuremap]
blindxfer=> ##
automon => *1
atxfer => *2

and I also added the options wW in the dial, howerver, when I call and
I press *1 nothing happen. Any ideas?



On 8/18/05, Matt Florell <[EMAIL PROTECTED]> wrote:
> Yes, there are different ways depending on how you use your system.
> The most unobtrusive way to the call is to use a computer app like a
> webpage or simple GUI app to send a Monitor command for the channel
> you are on(astguiclient does this: http://astguiclient.sf.net)
>  or
> If you are using CVS_Head from January 5 or later and you have it
> configured properly, you should just be able to hit *1 to begin
> recording, this will of course add a slight disconnection from the
> other person you are talking to, and it may not work if you are in a
> meetme conference, but it doesn't require a client computer to use:
> http://lists.digium.com/pipermail/asterisk-cvs/2005-January/004670.html
> 
> Hope that helps,
> 
> MATT---
> 
> 
> On 8/18/05, Eric <[EMAIL PROTECTED]> wrote:
> > Hi
> >
> > Is it possible to start recording a call during the conversation?
> >
> > --
> > Eric Smith
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Re: [Asterisk-Users] TAFM

[Il Neofita]

Hi,
I checked but I did not find any info regarding the config files. I
tough that I configured everything in the right way but I am not able
to see anything on the web page.
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[Asterisk-Users] TAFM

[Il Neofita]

Hi,
I installed this program but I am not able to configure, it does not
want to work.
Someone can help me?
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