[asterisk-users] Google Puts the Final Nail in the Google Voice Coffin

2014-04-09 Thread neo haux
No tell me that's a jock ! I can't believe it:

http://nerdvittles.com/?p=7940
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[asterisk-users] No voice when the calls come from Internet

2014-04-08 Thread neo haux
Hi,


 I have trouble establishing a call between between two SIP phones. One sip
phone is, with asterisk server, at home behind a firewall. The second sip
phone is an iPhone with 3G wireless connection.

 When I call from the SIP device at home the SIP account on the Internet
(iphone + 3G) I can hear voice on both devices. But when I call from the
Internet to my home SIP I get the ring but when I answer I don't hear any
thing!

 I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a
defined UDP range in rtp.conf forwarded to my Asterisk server.

 Do you have any idea when the voice is heard only when the call is from my
local network to the Internet and not in the other direction ?

 Nevertheless, when both SIP devices are in the same home IP network the
call is made without any problem whatever who starts the call.


Regards,
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[asterisk-users] (no subject)

2013-09-14 Thread neo haux
To Jonas:

I have an asterisk box at home and I have this line in my rtp.conf file:

rtpstart=1
rtpend=10100


And My FW is setup to forward all incoming ports of range 1-10100 to
the asterisk PC.
I've never had a problem since one year, but I have never received more
than two simultaneous calls with SIP clients.





Message: 5
Date: Fri, 13 Sep 2013 11:49:59 +0200
From: Jonas Kellens 
Subject: Re: [asterisk-users] RTP port ranges
To: Andrew Colin 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <5232dfc7.2030...@telenet.be>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Hello,

and when I define 11500 - 11954 it should use a random port in this range.

Where is it stated that you MUST use 1-2 ???

Someone else please ?


Jonas.


On 09/13/2013 11:46 AM, Andrew Colin wrote:
> Because normally it will use a random port between them
>
> On 9/13/2013 11:43 AM, Jonas Kellens wrote:
>> On 09/13/2013 11:41 AM, Andrew Colin wrote:
>>> Normally you should open ports 1-2 udp
>>>
>>>
>>>
>>> On 9/13/2013 11:37 AM, Jonas Kellens wrote:
 I now see that an IP-address gets blocked by my firewall because
 there are packets coming onto port 11955.
>>>
>>
>>
>> Why do I need such a big range ? That's like for 250 concurrent calls !
>>
>>
>>
>> Jonas.
>>
>
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[asterisk-users] How to use Skype ?

2013-09-02 Thread neo haux
Hi,

I want to recieve calls to my Skype account and forward them to a SIP/FXS
line. I searched for chan_skype for asterisk (v11), but found it only
available for asterisk 10

I know that Digium gives no support for this module, but I am sure that
someone somewhere did write some tool to allow such connectivity.

Do have any idea if I can use Skype with my asterisk v11 ?



Thanks
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[asterisk-users] Joining an astablished call

2013-05-05 Thread neo haux
Hi,

I don't know how to call this functionality, but what I want to do is join
an already established communication between PSTN---FXS_connected_phone
using my SIP phone (I have an asterisk v11 with digium TDM400P at home)

Is it possible? What I don't want is using the conference sound and
menu It's just a normal call between to channels that I have to  join
for few minutes.

Regards
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[asterisk-users] How to show caller number ?

2013-04-17 Thread neo haux
Hi,

I am using asterisk 11.1.0. How to display the caller number (from asterisk
-rvvv terminal) in the first step of the extension (before doing any
action) ?

Thanks
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[asterisk-users] weird "RED alarm" on FXO channel

2013-02-04 Thread neo haux
I have a recurrent problem on my asterisk box. I have "VIA Samuel 2" as a
CPU. With  asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled
from source.

I get a RED alrm drom the port 1( FXO) two or three times per day:

[Feb  4 15:54:57] WARNING[9991]: chan_dahdi.c:8018 handle_alarms: Detected
alarm on channel 1: Red Alarm




I plug the RJ11 in a normal phone and I can hear the line is up and I can
call extern number (via PSTN)

I must stop asterisk + restart dahdi and start asterisk to get asterisk
work normaly.   I have no idea where is the problem or even what to do to
debug the issue



I get these outputs when the channel is in "RED Alarm"

root@pbx01:~# /etc/init.d/dahdi status
### Span  1: WCTDM/0 "Wildcard TDM410P" (MASTER)
  1 FXOFXSKS   (In use) (EC: MG2 - INACTIVE)  RED  <
This is my problem !
  2 FXOFXSKS   (EC: MG2 - INACTIVE)  RED
  3 FXSFXOKS   (In use) (EC: MG2 - INACTIVE)
  4 FXSFXOKS   (In use) (EC: MG2 - INACTIVE)



pbx01*CLI> dahdi show status
Description  Alarms  IRQbpviol CRCFra
Codi Options  LBO
Wildcard TDM410P OK  0  0  0  CAS
Unk   0 db (CSU)/0-133 feet (DSX-1)
pbx01*CLI>


pbx01*CLI> dahdi show channels
   Chan Extension  Context Language   MOH InterpretBlocked
   State  Description
 pseudodefaultdefault
  In Service
  1from-pstn  default
  In Service
  3local  default
  In Service
  4local  default
  In Service
pbx01*CLI> dahdi show
cadences  channel   channels  statusversion
pbx01*CLI> dahdi show cadences
r1: 125,125,2000,4000
r2: 250,250,500,1000,250,250,500,4000
r3: 125,125,125,125,125,4000
r4: 1000,500,2500,5000



pbx01*CLI> dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC, MG2


pbx01*CLI> dahdi show channel 1
Channel: 1
Description:
File Descriptor: 12
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID:
Calling TON: 0
Caller ID subaddress:
Caller ID name: port-bell01
Mailbox: 1
Destroy: 0
InAlarm: 1
Signalling Type: FXS Kewlstart
Radio: 0
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Onhook




Is it a known bug ? Why I have to restart dhadi to resolve this ?

Thanks a lot !
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[asterisk-users] Build asterisk for VIA C3

2013-01-03 Thread neo haux
Is it difficult to publish a build asterisk.deb compiled for VIA
C3 architecture ? Instead of using the binary just for me.
So any one trying to install it on C3 CPU will need just to do:
aptitude install asterisk

The one that is installed by default doesn't work for such a CPU

Should I contact debian dev team for that?

Thanks


OLD messages ---
Message: 6
Date: Mon, 31 Dec 2012 12:08:40 -0500
From: neo haux 
Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
To: asterisk-users@lists.digium.com
Message-ID:

Content-Type: text/plain; charset=ISO-8859-1

Thanks George it works now !


Message: 6
Date: Sun, 30 Dec 2012 17:18:45 -0700
From: George Joseph 
Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset="utf-8"

Try this...  In menuselect, uncheck BUILD_NATIVE under Compiler Flags and
recompile.

On Sun, Dec 30, 2012 at 4:44 PM, neo haux  wrote:

> Hi,
>
> I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2
> 800MHz CPU. A small box to play with PBX at home.
>
> I get this error when I start asterisk:
>
> root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk  start
> Illegal instruction
> Starting Asterisk PBX: asteriskIllegal instruction
>
> I compiled it on debian 6.0.6 with this options:
>  ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \
> --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \
> --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib
>
>
> I was able last days to compile asterisk 1.8 and it did work
> perfectly except with Gtalk, and this is why I have to compile
> v11.1

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Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU

2012-12-31 Thread neo haux
Thanks George it works now !


Message: 6
Date: Sun, 30 Dec 2012 17:18:45 -0700
From: George Joseph 
Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset="utf-8"

Try this...  In menuselect, uncheck BUILD_NATIVE under Compiler Flags and
recompile.

On Sun, Dec 30, 2012 at 4:44 PM, neo haux  wrote:

> Hi,
>
> I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2
> 800MHz CPU. A small box to play with PBX at home.
>
> I get this error when I start asterisk:
>
> root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk  start
> Illegal instruction
> Starting Asterisk PBX: asteriskIllegal instruction
>
> I compiled it on debian 6.0.6 with this options:
>  ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \
> --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \
> --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib
>
>
> I was able last days to compile asterisk 1.8 and it did work
> perfectly except with Gtalk, and this is why I have to compile
> v11.1

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[asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU

2012-12-30 Thread neo haux
Hi,

I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2
800MHz CPU. A small box to play with PBX at home.

I get this error when I start asterisk:

root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk  start
Illegal instruction
Starting Asterisk PBX: asteriskIllegal instruction

I compiled it on debian 6.0.6 with this options:
 ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \
--with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \
--with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib


I was able last days to compile asterisk 1.8 and it did work
perfectly except with Gtalk, and this is why I have to compile
v11.1
I have gone through these steps (v1.8):

1) ./configure --with-iksemel=/usr/src/iksemel-1.4
--with-pri=/usr/src/libpri-1.4.14
--with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1'
--libdir=/usr/lib
2) in Makefile change : PROC=i586 (instead of i686)

3) in makeopts change:
#BUILD_PLATFORM=i686-pc-linux-gnu
BUILD_PLATFORM=i586-pc-linux-gnu
#BUILD_CPU=i686
BUILD_CPU=i586
#HOST_PLATFORM=i686-pc-linux-gnu
HOST_PLATFORM=i586-pc-linux-gnu
#HOST_CPU=i686
HOST_CPU=i586
PROC=i586

4) # make && make install

But I can't do the same changes for asterisk 11.1 because the
makefiles haven't the same content as for asterisk v1.8

I need asterisk 11.1 to be able to receive and make calls to gtalk accounts


Do you have any idea where is the problem ?



My VIA CPU:

root@pbx01:/usr/src/asterisk-11.1.0# cat /proc/cpuinfo
processor   : 0
vendor_id   : CentaurHauls
cpu family  : 6
model   : 7
model name  : VIA Samuel 2
stepping: 3
cpu MHz : 799.892
cache size  : 64 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu de tsc msr cx8 mtrr pge mmx 3dnow
bogomips: 1599.78
clflush size: 32
cache_alignment : 32
address sizes   : 32 bits physical, 32 bits virtual
power management:

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[asterisk-users] version compatible with centos 5.7 (2.6.18-308.8.2.el5)

2012-08-21 Thread neo nortan

dear guys
plz tell me which version of asterisk is compatible with centos 5.7 
(2.6.18-308.8.2.el5).
and which is the latest version.

regards
neo
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[asterisk-users] freepbx asterisk

2012-07-20 Thread neo nortan

dear
i am a neewbi for asterisk, plz tell me or if any link is there where i can 
understand how asterisk, freepbx, web-meetme, dahdi all these tools works and 
how they are related.
plz help me.

regards

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[asterisk-users] Fwd: asterisk-users Digest, Vol 95, Issue 33

2012-06-24 Thread neo haux
Thanks I had this line in my /etc/asterisk/chan_dahdi.conf :

include=/etc/asterisk/dahdi-channels.conf


the file /etc/asterisk/dahdi-channels.conf was generated by
/usr/sbin/dahdi_genconf

I simply did that :
cat /etc/asterisk/dahdi-channels.conf >> /etc/asterisk/chan_dahdi.conf

It works now.

May be the option "include" is not supported within the file chan_dahdi.conf

--

Message: 2
Date: Sat, 23 Jun 2012 16:34:35 -1000
From: Julian Yap 
Subject: Re: [asterisk-users] Can't make call with TDM410P
To: Asterisk Users Mailing List - Non-Commercial Discussion
       
Message-ID:
       
Content-Type: text/plain; charset=ISO-8859-1

On Sat, Jun 23, 2012 at 10:32 AM, neo haux  wrote:
> Actually I can start and receive SIP calls (PC client, iphone client)
> but?I have an issue with calling external number throught PSTN
> (certified-asterisk-1.8.11-cert2).
>
> I'm having this ?error when making a call:
>
> *CLI> ? == Using SIP RTP CoS mark 5
> ? ? -- Executing [9000@local:1] Dial("SIP/3000-0006",
> "DAHDI/1/4384019357,10") in new stack
> [Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
> ? == Everyone is busy/congested at this time (1:0/0/1)
> ? ? -- Executing [9000@local:2] Hangup("SIP/3000-0006", "") in new stack
> ? == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-0006'
>
>
> My configs :
> *CLI> dahdi show channels
> ? ?Chan Extension ?Context ? ? ? ? Language ? MOH Interpret
> Blocked ? ?State
> ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default
> ?? ? ? In Service

Where are your channels?  That's why you are receiving the error
"Unable to create channel of type 'DAHDI'".

Define your channel groups in /etc/asterisk/chan_dahdi.conf

Then is should look like this:
# asterisk -rx "dahdi show channels"
  Chan Extension  Context         Language   MOH Interpret
Blocked    State
 pseudo            default                    default
      In Service
     1            from-external   en         default
      In Service
     2            from-external   en         default
      In Service
     3            from-external   en         default
      In Service
     4            from-external   en         default
      In Service
     5            from-external   en         default
      In Service
     6            from-external   en         default
      In Service
...

- Julian

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[asterisk-users] Can't make call with TDM410P

2012-06-23 Thread neo haux
Actually I can start and receive SIP calls (PC client, iphone client)
but I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2).

I'm having this  error when making a call:

*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [9000@local:1] Dial("SIP/3000-0006",
"DAHDI/1/4384019357,10") in new stack
[Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [9000@local:2] Hangup("SIP/3000-0006", "") in new stack
  == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-0006'


My configs :
*CLI> dahdi show channels
   Chan Extension  Context         Language   MOH Interpret
Blocked    State
 pseudo            default                    default
       In Service
*CLI> dahdi show status
Description                              Alarms  IRQ    bpviol CRC
Fra Codi Options  LBO
Wildcard TDM410P                         OK      0      0      0
CAS Unk           0 db (CSU)/0-133 feet (DSX-1)
*CLI>


root@my-PC:/usr/src/certified-asterisk-1.8.11-cert2# lsmod | grep dahdi
dahdi_echocan_mg2      12998  4
dahdi_voicebus         58608  1 wctdm24xxp
dahdi                 220595  3 dahdi_echocan_mg2,wctdm24xxp,dahdi_voicebus
crc_ccitt              12667  2 wctdm24xxp,dah




extensions.conf
[local]
exten => 100,1,Dial(gtalk/asterisk/myaccount.v...@gmail.com)
exten => 2000,1,Dial(SIP/2000,10)
exten => 3000,1,Dial(SIP/3000,10)
exten => 9000,1,Dial(DAHDI/1/MyCellPhoneNumber,10)
exten => 9000,2,hangup()



root@My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_scan
[1]
active=yes
alarms=OK
description=Wildcard TDM410P
name=WCTDM/0
manufacturer=Digium
devicetype=Wildcard TDM410P
location=PCI Bus 04 Slot 01
basechan=1
totchans=4
irq=0
type=analog
port=1,FXO
port=2,FXO
port=3,FXS
port=4,FXS

root@My-PC:/usr/src/certified-asterisk-1.8.11-cert2# dahdi_hardware
pci::04:00.0 wctdm24xxp+  d161:8005 Wildcard TDM410P

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Re: [asterisk-users] Delay before ringing from PSTN`s call

2011-10-05 Thread neo haux
Hi,

I commented the option callerid in the file dahdi-channels.conf without
success, My SIP phone still ring after 4-5 secondes :-(

; Span 1: WCTDM/0 "Wildcard TDM410P" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)"
signalling=fxs_ks
;callerid=asreceived

I am living in Canada, so I guess I should use USA signaling ? If so in
which file ?



Message: 7
Date: Tue, 04 Oct 2011 14:49:55 -0400
From: John Novack 
Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call
To: Asterisk Users Mailing List - Non-Commercial Discussion
   
Cc: neo haux 
Message-ID: <4e8b5553.2030...@stromberg-carlson.org>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your
country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper
configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protocols
You will need to dig into that first.

John Novack


neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
>
> I configured incoming calls from pstn to ring my SIP phone (extension :
100)
>
> cat  extensions.conf
> ...
> [from-pstn]
> exten => s,1,Dial(SIP/100,10)
>  same => n,VoiceMail(100,u)
>
>
>
>
> root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
> ...
> ...
> ...
> ;;; line="1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)"
> signalling=fxs_ks
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
> ...
> ...
> ...
>
> What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.
>
> I did those modifications in the file  /etc/asterisk/chan_dahdi.conf
without improuvement ( After restarting Asterisk)
>
> [channels]
> cidstart=ring
> immediate=yes
> faxdetect=no
> usecallerid=no
>
>
> Here is the debug from Asterisk console
>
> *CLI> -- Starting simple switch on 'DAHDI/1-1'
> -- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new
stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/100
> -- SIP/100-0001 is ringing
>   == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
> -- Hanging up on 'DAHDI/1-1'
> -- Hungup 'DAHDI/1-1'
>
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[asterisk-users] Delay before ringing from PSTN`s call

2011-10-03 Thread neo haux
Hi

I am testing a degium TDP400P (2fxo+2fxs) on my asterisk

I configured incoming calls from pstn to ring my SIP phone (extension : 100)

cat  extensions.conf
...
[from-pstn]
exten => s,1,Dial(SIP/100,10)
 same => n,VoiceMail(100,u)




root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
...
...
...
;;; line="1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default
...
...
...

What I don`t understand is why the SIPphone rings after 3 secondes after
Astererisk detects the incoming call. Moreover, after hanging off the
external caller the SIPphone continue to ring for 3 seconds.

I did those modifications in the file  /etc/asterisk/chan_dahdi.conf without
improuvement ( After restarting Asterisk)

[channels]
cidstart=ring
immediate=yes
faxdetect=no
usecallerid=no




Here is the debug from Asterisk console

*CLI> -- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:1] Dial("DAHDI/1-1", "SIP/100,10") in new
stack
  == Using SIP RTP CoS mark 5
-- Called SIP/100
-- SIP/100-0001 is ringing
  == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
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Re: [asterisk-users] RTP stream when * and Xlite are both behind Nat devices.

2011-09-22 Thread neo haux
Hi rcswebb,


I had a problem like yours :
Asterisk -NAT - internet - NAT - 3CX phone

Without modifiyng Astrisk conf I could start a call from the client but
without hearing a sound.

The solution for me was to force Asterisk to modify the outgoing udp packet
to insert it's public ip and not the private IP behind the NAT .

So in your sip.conf I modified :

[general]
*externip=YouEternalIP*
NAT=Yes


Hope that'll help :-)

Message: 11
Date: Wed, 21 Sep 2011 10:52:08 +0100
From: Richard Webb 
Subject: [asterisk-users] RTP stream when * and Xlite are both behind
   Nat devices.
To: asterisk-users@lists.digium.com
Message-ID:
   
Content-Type: text/plain; charset="iso-8859-1"

Hi,

I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone
behind a different NAT network.

Asterisk -> Nat -> Internet -> Nat -> Softphone.

I can register my softphone to the asterisk box ok via SIP but the RTP
stream from the asterisk box is addressed to the private non-routeable
address of the softphone when I turn on rtp debuging.

How can I configure the rtp stream to be sent to the public facing address
of the softphone?

Cheers,
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Re: [asterisk-users] Phone numbers and asterisk

2011-09-05 Thread neo haux
I mean the directory of phone numbers is stored within asterisk. So the SIP
phone just fetch that list when it starts.


--

Message: 3
Date: Sun, 4 Sep 2011 19:47:00 -0400
From: Robert-iPhone 
Subject: Re: [asterisk-users] Phone numbers and asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
   
Message-ID: <0d19da7a-0766-4ae9-a967-528ccae36...@gmail.com>
Content-Type: text/plain; charset="us-ascii"

what do you mean? Like speed dial or directory?

Sent from my iPhone

On Sep 4, 2011, at 6:47 PM, neo haux  wrote:

> Hi,
>
> It is possible to save all the phones numbers on asterisk servers instead
of doing so manually in each VoIP device ?
>
> Does SIP take care of such configuration ?
>
> Thanks
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[asterisk-users] Phone numbers and asterisk

2011-09-04 Thread neo haux
Hi,

It is possible to save all the phones numbers on asterisk servers instead of
doing so manually in each VoIP device ?

Does SIP take care of such configuration ?

Thanks
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[asterisk-users] Asterisk+internal phones+recorded messages

2011-08-10 Thread neo haux
Hi

I want to change my old answering phone machine and two wireless phones with
asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel
9133i) + Wifi/SIP phone

I am wondering if I´ll lost actual functionalities that are present in my
old answering machine:
1) is it possible to show the caller number (coming from PSTN/FXO) in both
SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this
functionality

2) Most important question is : can I see on those internal phones (Wifi/SIP
phone  and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I
have this fucntionality with my old answering machine where I can see the
number of new messages recorded in a big LCD screen.


Thx
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Re: [asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)

2011-08-06 Thread neo haux
Hi Asterisk lovers,

Thank you very much Warren and  Tzafrir ! I resolved the issu by
installing openssl-dev.



Date: Tue, 2 Aug 2011 15:40:17 -0500
From: Warren Selby 
Subject: Re: [asterisk-users] Problem with
   (asterisk1.8-iksemel1.4-GoogleVoice)
To: Asterisk Users Mailing List - Non-Commercial Discussion
   
Message-ID: <7648899e-b872-4d34-91a0-e67c73061...@selbytech.com>
Content-Type: text/plain;   charset=utf-8

Install OpenSSL-devel (or whatever the equivalent ubuntu package is
called) and then recompile / reinstall and test it again.

Thanks,
--Warren Selby, dCAP

On Aug 2, 2011, at 12:06 PM, neo haux  wrote:

> Hi,
>
> I?ve compiled asterisk-1.8.5.0   on my Debian based distro (Pinguy)
> I also compiled iksemel (v1.4) with the option 2./configure
> --with-libgnutls-prefix=/usr"
> As explained in this link (to avoid compilation error )
> http://code.google.com/p/iksemel/issues/detail?id=29#c3
>
> I configured jabber.conf and gtalk.conf as explained in
> wiki.asterisk.org, but I have this error when starting :
> asterisk -c

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[asterisk-users] Problem with (asterisk1.8-iksemel1.4-GoogleVoice)

2011-08-02 Thread neo haux
Hi,

I´ve compiled asterisk-1.8.5.0   on my Debian based distro (Pinguy)
I also compiled iksemel (v1.4) with the option 2./configure
--with-libgnutls-prefix=/usr"
As explained in this link (to avoid compilation error )
http://code.google.com/p/iksemel/issues/detail?id=29#c3

I configured jabber.conf and gtalk.conf as explained in
wiki.asterisk.org, but I have this error when starting :
asterisk -c
...
...
JABBER: asterisk INCOMING: http://etherx.jabber.org/streams";
xmlns="jabber:client">X-GOOGLE-TOKENX-OAUTH2
[Aug  2 12:57:40] ERROR[31479]: res_jabber.c:1610 aji_act_hook:
OpenSSL not installed. You need to install OpenSSL on this system, or
disable the TLS option in your configuration file
[Aug  2 12:57:40] WARNING[31479]: res_jabber.c:1391 aji_recv: Parsing
failure: Hook returned an error.
[Aug  2 12:57:40] WARNING[31479]: res_jabber.c:2742 aji_recv_loop:
JABBER: Got hook event.
[Aug  2 12:57:56] WARNING[31479]: res_jabber.c:2753 aji_recv_loop:
JABBER: socket read error


But I have already openssl :
root@mohamed-desktop:/usr/src# dpkg -l |grep -i openssl
ii  libcurl3           7.21.0-1ubuntu1.1
Multi-protocol file transfer library (OpenSSL)
ii  libxmlsec1-openss  1.2.14-1+squeeze1build0.10.10.1       Openssl
engine for the XML security library
ii  openssl            0.9.8o-1ubuntu4.4                     Secure
Socket Layer (SSL) binary and related cryptographic tools
ii  python-op          0.10-1                                Python
wrapper around the OpenSSL library
ii  ssl-cert           1.0.26                                simple
debconf wrapper for OpenSSL


Have you any idea where is the problem ?
NB: I didn´t have that problem with asterisk 1.6

Thx

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Re: [asterisk-users] Asterisk crashes : Failed to start PBX

2009-11-20 Thread Neo Anderson
Hi,

I have already setup to rotate logs hourly & Debug level is 3.
Is there any other possibility of crash?

Thanks in advance!!


--
Regards,
Voipexpert






From: Giorgio Incantalupo 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Fri, November 20, 2009 10:06:06 PM
Subject: Re: [asterisk-users] Asterisk crashes : Failed to start PBX

Hi Neo,

have you checked your log files? It sometimes happened to me that 
Asterisk crashed without a reason. I discovered my logrotate didn't make 
its dirty work so I had huge log files. I lowered Asterisk log level and 
forced logrotate to work and now I have no more crashes.

Hope it may help. :)

Giorgio.

Neo Anderson wrote:
> Hello,
>
> I am using Asterisk 1.4.24.1 version in production.
> OS is Centos 5.3 64 bit & RAM is 8 GB.
> I am facing crash in asterisk approx each 12 hour.
> When it crashes I see  below lines in asterisk logs.
> [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
> [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
> I debugged asterisk source code in details &  I found that it happens 
> because it can not allocate memory to create thread.
>
> Another thing is, when I check coredump using gdb, it's not showing 
> any debug symbols.
>
> Would you please let me know how to prevent or resolve this?
>
> Thanks in advance!!
>
> --
> Regards,
> voipexpert
>
>
> 
>
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[asterisk-users] Asterisk crashes : Failed to start PBX

2009-11-18 Thread Neo Anderson
Hello,

I am using Asterisk 1.4.24.1 version in production.
OS is Centos 5.3 64 bit & RAM is 8 GB.
I am facing crash in asterisk approx each 12 hour.
When it crashes I see  below linesin asterisk logs.
[Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread
[Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(
I debugged asterisk source code in details &  I found that it happens because 
it can not allocate memory to create thread.

Another thing is, when I check coredump using gdb, it's not showing any debug 
symbols.

Would you please let me know how to prevent or resolve this?

Thanks in advance!!

--
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voipexpert


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[Asterisk-Users] Authentication Problem

2005-01-20 Thread Neo
Hello everbody,
I am having problems is Database version and Real time version of Asterisk.
Users are connecting with no problem,
they gets authenticate and its working fine,
but
after 2-3 minutes, registration with the same user comes and it gets 
failed to authenticate. dial tone gone, users unable to call,

but this behaviour not remains for the all users for all the time.
most of the time they are able to call,
its totally wiered to me.
any ideas ?
-Neo
p.s
i m using different kinds of clients
xlite
xpro
cisco ata
dta
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Re: [Asterisk-Users] Alcatel PBX

2004-11-19 Thread neo
hey Torsten,

i tried overlapdial=yes and guess what ?
IT WORKED 

i owe you one.

thanks friend.

-Neo

>Hello,

>On Fri, 19 Nov 2004, pbx wrote:

> [EMAIL PROTECTED] wrote:
>
> >i have the following scnario.
> >
> >1. Alcatel PBX with e1 module
> >2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1
> >connected to alcatel pbx.
> >
> >i m having problem in outgoing from alcatel.
> >incoming from pstn -> asterisk -> alcatel working fine, but outgoing from
> >alcatel -> asterisk -> pstn or any sip extensions not working. it hangs up
the
> >line as soon as i answer the call. i have generated dialtone via playtones
but
> >it has also issue.
> >
> >when i connect pstn e1 line directly to altacel e1 module, it works fine, but
> >behind asterisk it hangups.
> >
> >any body have good idea ?

Did you set overlapdial=yes in your zapata.conf?


> >
> >further details can be provided if u need more.
> >
> >regards.
> >-Neo
> >
> >
> >
> >
> >This message was sent using IMP, the Internet Messaging Program.
> >
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> >
> >
> >
> It might be a good Idea, to at least send your config file zaptel,
> zapata,and extension
> and  the message  that you got on the console  when  the problem occurs
> ... Regards,
> Jack
>
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This message was sent using IMP, the Internet Messaging Program.

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[Asterisk-Users] Alcatel PBX

2004-11-19 Thread neo


Dear Users,

i have the following scnario.

1. Alcatel PBX with e1 module
2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1
connected to alcatel pbx.

i m having problem in outgoing from alcatel.
incoming from pstn -> asterisk -> alcatel working fine, but outgoing from
alcatel -> asterisk -> pstn or any sip extensions not working. it hangs up the
line as soon as i answer the call. i have generated dialtone via playtones but
it has also issue.

when i connect pstn e1 line directly to altacel e1 module, it works fine, but
behind asterisk it hangups.

any body have good idea ?

further details can be provided if u need more.

regards.
-Neo




This message was sent using IMP, the Internet Messaging Program.

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[Asterisk-Users] why ata stop working after 10 mins after registering from mysql

2004-07-14 Thread neo


hello,

i m using sip.conf and extension.conf from mysqldb

my problem is after 10 mins of registration ata stops working, it works fine 
for outgoing but fail for incoming.

any help ?

-neo
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[Asterisk-Users] Voange with asterisk settings

2004-07-14 Thread neo

hello, 

> here comes the settings.
> sip.conf
> register => username:[EMAIL PROTECTED]:5061/1010
> [vonage]
> type=friend
> username=yourusername
> secret=yourpw
> host=sphone.vopr.vonage.net
> port=5061
> disallow=all
> allow=ulaw
> maxexpirey=15
> dtmfmode=inband
> fromuser=yourusername
> fromdomain=sphone.vopr.vonage.net
> canreinvite=no
> nat=yes
> 
> in extensions.conf
> 
> exten => _1.,1,Dial(SIP/[EMAIL PROTECTED],1000,tr)
 
> for inbound in exten.conf
> exten => _yourusername,1,whateveryouwanttodo

 enjoy guys :)
 
 let me know if u feel prob

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[Asterisk-Users] Vonage working with asterisk

2004-07-14 Thread neo
atlast after working of 7 hours i got voange soft account working on asterisk.


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[Asterisk-Users] warning message (sound card) - when I run asterisk!!!

2004-05-27 Thread Neo Jia
All,
   After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:

chan_oss.so] => (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
  == Registered channel type 'Console' (OSS Console
Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238
sound_thread: Read error on sound device: Resource
temporarily unavailable
 [chan_phone.so] => (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux
Telephony API Driver)

My sound card information:

Vendor : Intel Corp.
Model  : 82801CA/CAM AC'97 Audio Controller
Module : i810_audio

After running 'dial' command under the asterisk
prompt, I got the following message without any sound.

*CLI> -- Executing Wait("OSS/dsp", "1") in new
stack
-- Executing Answer("OSS/dsp", "") in new stack
 << Console call has been answered >>
-- Executing DigitTimeout("OSS/dsp", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("OSS/dsp", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("OSS/dsp",
"demo-congrats") in new stack
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408
soundcard_setinput: Unable to re-open DSP device:
Device or resource busy
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567
oss_write: Unable to set device to input mode
May 26 00:40:55 WARNING[-1221268560]: file.c:537
ast_readaudio_callback: Failed to write frame
-- Playing 'demo-congrats' (language 'en')
  == Spawn extension (local, s, 5) exited non-zero on
'OSS/dsp'

Is there anyone can give me any hints or help?

Thanks,
Neo

[Asterisk-Users] warning message (sound card) - when I run asterisk!!!

2004-05-26 Thread Neo Jia





  
  
All,
   After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:

chan_oss.so] => (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
  == Registered channel type 'Console' (OSS Console
Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
May 26 00:37:58 WARNING[-1168819280]: chan_oss.c:238
sound_thread: Read error on sound device: Resource
temporarily unavailable
 [chan_phone.so] => (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux
Telephony API Driver)

My sound card information:

Vendor : Intel Corp.
Model  : 82801CA/CAM AC'97 Audio Controller
Module : i810_audio

After running 'dial' command under the asterisk
prompt, I got the following message without any sound.

*CLI> -- Executing Wait("OSS/dsp", "1") in new
stack
-- Executing Answer("OSS/dsp", "") in new stack
 << Console call has been answered >>
-- Executing DigitTimeout("OSS/dsp", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("OSS/dsp", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("OSS/dsp",
"demo-congrats") in new stack
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408
soundcard_setinput: Unable to re-open DSP device:
Device or resource busy
May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567
oss_write: Unable to set device to input mode
May 26 00:40:55 WARNING[-1221268560]: file.c:537
ast_readaudio_callback: Failed to write frame
-- Playing 'demo-congrats' (language 'en')
  == Spawn extension (local, s, 5) exited non-zero on
'OSS/dsp'

Is there anyone can give me any hints or help?

Thanks,
Neo



[Asterisk-Users] vonage sip url

2004-05-04 Thread neo
Hello List,

anybody knows the sip url of vonage ???

like [EMAIL PROTECTED] ??

regards.
-Neo
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[Asterisk-Users] Zap Outgoing

2004-04-19 Thread neo
Hello All,

i m having busy signal when i dial any number, while incoming on zap is working 
fine and its transfering to my soft phone.

some time back outgoing was working ok but now i dont know what i messed up.

any idea ?

it gives busy signal after Zap/25-1 answered SIP/300


-Neo

= Spawn extension (voicepulse-incoming, s, 1) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Executing Dial("SIP/3000-2e72", "Zap/25/18005558355") in new stack
-- Called 25/18005558355
-- Zap/25-1 answered SIP/3000-2e72

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[Asterisk-Users] Incoming on Zap

2004-04-18 Thread neo
Hello Everybody,

I have 1 X100P (FXO) card attached to my *, also there is a line connected to 
it, but when i dial that number it is not forwarding to anywhere, just * 
recognizes that the call is coming.

I want incoming call to be forward on my x-lite extension lets say 2000,

can anybody tell me the settings of extensions.conf and other conf files.

thanks and best regards.
-Neo
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[Asterisk-Users] DLINK DPH-70 with asterisk

2004-04-16 Thread neo


Hello everybody,

i have DLINK DPH-70 Phone,

does anybody know if it works with asterisk ?

i have g729 codec installed on my asterisk server which the phone supports, and 
it gets authenticate with asterisk but when i make a call it says maximum tries 
reaches for dialing..

regards.
-neo


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