from:"Marco"


pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.

That way would be easier to help you.

On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:

Hello,

I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :

[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes

#include iax.voip1.conf
#include iax.renoir.conf

The iax.voip1.conf file contains :

[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes


The iax.renoir.conf file contains :

[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm

Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).

Anyone who has any idea ?
Thanks,
jb


PS :
(The debug in the .184 machine :

   -- Accepting UNAUTHENTICATED call from 10.0.0.160:
 requested format = ulaw,
 requested prefs = (alaw),
 actual format = gsm,
 host prefs = (gsm),
 priority = mine
 -- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
 -- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack

with the following extensions.conf :

[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})

[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)


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Marco Mouta
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Re: [asterisk-users] Iax bug ?


Hi,

Unfortunately i'm not able to debug this with you now :( I'm busy.

[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=

This secret empty is this allowed?

inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes

Try a simple test with this, and then step by step go to rsa authentication.

http://astrecipes.net/index.php?n=204

If in troubles, post here i'll try to help you

By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.



On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:

Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality...  Mr. Roboto.
allow=alaw  ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes

[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all

How .160 call .184 :

exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})

How .184 call .160 :

exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)

Thanks,
jb


Marco Mouta a écrit :
 pls post iax.conf of Both machines , as well as your dial() string on
 both servers to connect each other.

 That way would be easier to help you.

 On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
 Hello,

 I'm french, so excuse my poor English.
 I'm face to a terrible thing, with has stole a lot of my time.
 On the .184 machine, I've the following iax.conf :

 [general]
 rtcachefriends=yes
 bandwidth=high
 tos=reliability
 jitterbuffer=no
 autokill=yes

 #include iax.voip1.conf
 #include iax.renoir.conf

 The iax.voip1.conf file contains :

 [VOIP1]
 type=friend
 host=10.0.0.160
 auth=rsa
 secret=
 inkey=voip3
 outkey=voip1
 context=CONTEXT_VOIP1
 allow=all
 ipaddr=10.0.0.160
 port=4569
 qualify=yes
 trunk=yes


 The iax.renoir.conf file contains :

 [VOIP_RENOIR]
 type=friend
 host=renoir.lucyde
 auth=rsa
 inkey=key_184
 outkey=key_Renoir
 context=CONTEXT_RENOIR
 trunk=yes
 allow=gsm

 Thanks to the variable context, when .184 receive a call from .160, this
 call should be executed in the CONTEXT_VOIP1. In fact the call is
 executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
 the call is executed in the context of the last section's context of the
 iax.conf file (e.g. CONTEXT_RENOIR here).

 Anyone who has any idea ?
 Thanks,
 jb


 PS :
 (The debug in the .184 machine :

-- Accepting UNAUTHENTICATED call from 10.0.0.160:
  requested format = ulaw,
  requested prefs = (alaw),
  actual format = gsm,
  host prefs = (gsm),
  priority = mine
  -- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
 CONTEXT_RENOIR) in new stack
  -- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
 in new stack

 with the following extensions.conf :

 [CONTEXT_VOIP1]
 exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
 exten = _X.,2,Macro(check_forward,${EXTEN})

 [CONTEXT_RENOIR]
 exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
 exten = _X.,2,Macro(check_forward,${EXTEN})
 )


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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
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Marco Mouta
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Re: [asterisk-users] Iax bug ?


Hi,

I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:

Take a look on incoming call authentication, and how asterisk handles this:

http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication

Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.

If a username is supplied, Asterisk does the following:

   * Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
   * If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
   * Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
   * Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.

If a username is not supplied, Asterisk does the following:

   * Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
   * Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
   * Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
   * Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.


Hope this helps!

I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:

Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.


Pls give some feedback if you solved the problem.









On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:

Hi,

Unfortunately i'm not able to debug this with you now :( I'm busy.

[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes

Try a simple test with this, and then step by step go to rsa authentication.

http://astrecipes.net/index.php?n=204

If in troubles, post here i'll try to help you

By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.



On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
 Here the .160's iax.conf file :
 [general]
 bandwidth=high
 tos=reliability
 bandwidth=low
 disallow=all; Icky sound quality...  Mr. Roboto.
 allow=alaw  ; Always allow GSM, it's cool :)
 jitterbuffer=no
 forcejitterbuffer=no
 tos=lowdelay
 autokill=yes

 [VOIP1]
 type=friend
 host=10.0.0.184
 auth=rsa
 inkey=voip3
 outkey=voip1
 context=VOIPLINK3
 qualify=1
 trunk=yes
 allow=all

 How .160 call .184 :

 exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})

 How .184 call .160 :

 exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
 (the same)

 Thanks,
 jb


 Marco Mouta a écrit :
  pls post iax.conf of Both machines , as well as your dial() string on
  both servers to connect each other.
 
  That way would be easier to help you.
 
  On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
  Hello,
 
  I'm french, so excuse my poor English.
  I'm face to a terrible thing, with has stole a lot of my time.
  On the .184 machine, I've the following iax.conf :
 
  [general]
  rtcachefriends=yes
  bandwidth=high
  tos=reliability
  jitterbuffer=no
  autokill=yes
 
  #include iax.voip1.conf
  #include iax.renoir.conf
 
  The iax.voip1.conf file contains :
 
  [VOIP1]
  type=friend
  host=10.0.0.160
  auth=rsa
  secret=
  inkey=voip3
  outkey=voip1
  context=CONTEXT_VOIP1
  allow=all
  ipaddr=10.0.0.160
  port=4569
  qualify=yes
  trunk

Re: [asterisk-users] Direct call vs Block call


pls post your misdn.conf as well as extensions.conf, so someone could
help you on this.

On 10/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:

Hi for all,

 i 've installed asterisk with isdn trunk with alcatel pabx.
 For alcatel users use asterisk lines, should dial 0 to take tone from
asterisk. In default configuration in alcatel, if user dial 0 this error
occour:

 !! Unexpected Channel selection 3
 -- Extension '' in context 'default' from '' does not exist. Rejecting call
on channel 0/31, span 1

 In alcatel we're enable block dial, so alcatel only send to asterisk when
user end dialing all digits.

 How i permit the first case to work ??

 Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Marco Mouta
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Re: [asterisk-users] asterisk misdn incoming line not working.


Plse Read bellow:

On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:

Hi list,

I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.

when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.

P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
extension
P[ 1] MGMT: SSTATUS: L1_ACTIVATED
  == Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
  == Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
context 'default'
Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
sent into invalid extension 's' in context 'default', but no invalid handler

087822291 is the number i dial from, 0594643637 is the number that the
asterisk server should respond to.

in misdn.conf i created a kpn section like this:
[kpn]
ports=1ptmp
;group=1
immediate=yes
always_immediate=yes
context=kpn-in
hold_allowed=yes
msns=*

and in extensions.conf i created a very basic kpn-in section like this:
[kpn-in]

HERE IS your PROBLEM:


exten = kpn-in,1,Dial(SIP/mark,25,tr)


1- Be sure of of MSNs string your telco is sending you.
2- Do this:

[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup


reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.

Hope this helps,

Pls. give me some feedback.






I don't really have much experience with asterisk so I probably did something
wrong here, but I couldn't really figure out how to get it done.

anyone out there any ideas?

Mark
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Re: [asterisk-users] asterisk misdn incoming line not working.


My mistake:


[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup



On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:

Plse Read bellow:

On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
 Hi list,

 I have a server running a simple hfs isdn card running with chan_misdn.
 the problem is, I can't get asterisk to pick up the phone, outgoing calls work
 fine.

 when running asterisk with -vvvc I get the following log when I try
 to dial the isdn server.

 P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
 extension
 P[ 1] MGMT: SSTATUS: L1_ACTIVATED
   == Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
   == Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
 context 'default'
 Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
 sent into invalid extension 's' in context 'default', but no invalid handler

 087822291 is the number i dial from, 0594643637 is the number that the
 asterisk server should respond to.

 in misdn.conf i created a kpn section like this:
 [kpn]
 ports=1ptmp
 ;group=1
 immediate=yes
 always_immediate=yes
 context=kpn-in
 hold_allowed=yes
 msns=*

 and in extensions.conf i created a very basic kpn-in section like this:
 [kpn-in]
HERE IS your PROBLEM:

 exten = kpn-in,1,Dial(SIP/mark,25,tr)

1- Be sure of of MSNs string your telco is sending you.
2- Do this:

[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup


reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.

Hope this helps,

Pls. give me some feedback.





 I don't really have much experience with asterisk so I probably did something
 wrong here, but I couldn't really figure out how to get it done.

 anyone out there any ideas?

 Mark
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Com os melhores cumprimentos,

Marco Mouta




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Re: [asterisk-users] Iax bug ?


Why r u using rsa authentication? you should start with something
simple. test the link i sent u.

On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:

Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
jb

Marco Mouta a écrit :
 Hi,

 I think i found your problem, look that in your debug you have, -
 Accepting UNAUTHENTICATED call from 10.0.0.160:

 Take a look on incoming call authentication, and how asterisk handles this:

 http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication

 Incoming Connections
 When Asterisk receives an incoming IAX connection, the initial call
 information can include a username (in the IAX2 USERNAME field) or
 not. In addition, the incoming connection has a source IP address that
 Asterisk can use for authentication as well.

 If a username is supplied, Asterisk does the following:

* Search iax.conf for a type=user entry with a section name (eg
 [username]) matching the supplied username; if no matching entry is
 found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
 IP address of the caller to these lists. If the connection is not
 allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
 it fails, refuse the connection.
* Accept the connection and send the caller to the context
 specified in the context setting for this iax.conf entry.

 If a username is not supplied, Asterisk does the following:

* Search for a type=user entry in iax.conf with no secret
 specified and also allow and/or deny restrictions that do not restrict
 the caller from connecting. If such an entry is found, accept the
 connection, and use the name of the found iax.conf entry as the
 connecting username.
* Search for a type=user entry in iax.conf with no secret
 specified and no allow and/or deny restrictions at all. If such an
 entry is found, accept the connection. and use the name of the found
 iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
 key) specified and also allow and/or deny restrictions that do not
 restrict the caller from connecting. If such an entry is found,
 attempt to authenticate the caller using the specified secret or key,
 and if that passes, accept the connection, and use the name of the
 found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
 key) specified and no allow and/or deny restrictions at all. If such
 an entry is found, attempt to authenticate the caller using the
 specified secret or key, and if that passes, accept the connection,
 and use the name of the found iax.conf entry as the connecting
 username.


 Hope this helps!

 I didn't read all, but what i guess is: the incoming call isn't being
 correctly authenticated, so can't go to VOIP1 as you desire, then as
 is mention above:

 Search for a type=user entry in iax.conf with no secret specified
 and no allow and/or deny restrictions at all. If such an entry is
 found, accept the connection. and use the name of the found iax.conf
 entry as the connecting username.


 Pls give some feedback if you solved the problem.









 On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
 Hi,

 Unfortunately i'm not able to debug this with you now :( I'm busy.

 [VOIP1]
 type=friend
 host=10.0.0.160
 auth=rsa
 secret=
 This secret empty is this allowed?
 inkey=voip3
 outkey=voip1
 context=CONTEXT_VOIP1
 allow=all
 ipaddr=10.0.0.160
 port=4569
 qualify=yes
 trunk=yes

 Try a simple test with this, and then step by step go to rsa
 authentication.

 http://astrecipes.net/index.php?n=204

 If in troubles, post here i'll try to help you

 By the way, to understand much better what's going on i would
 recommend you to not use type=friend and use type=user and type=peer.



 On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
  Here the .160's iax.conf file :
  [general]
  bandwidth=high
  tos=reliability
  bandwidth=low
  disallow=all; Icky sound quality...  Mr. Roboto.
  allow=alaw  ; Always allow GSM, it's cool :)
  jitterbuffer=no
  forcejitterbuffer=no
  tos=lowdelay
  autokill=yes
 
  [VOIP1]
  type=friend
  host=10.0.0.184
  auth=rsa
  inkey=voip3
  outkey=voip1
  context=VOIPLINK3
  qualify=1
  trunk=yes
  allow=all
 
  How .160 call .184 :
 
  exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
 
  How .184 call .160 :
 
  exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
  (the same)
 
  Thanks,
  jb
 
 
  Marco Mouta a écrit :
   pls post iax.conf of Both machines , as well as your dial() string on
   both servers to connect each other.
  
   That way would be easier to help you.
  
   On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
   Hello,
  
   I'm french, so excuse my poor English.
   I'm face to a terrible thing, with has stole a lot

[asterisk-users] CDR_DISPOSITION_FAILED - Call has been answered correctly

2006-10-24 Thread Marco Mouta


Hi guys,

I've an asterisk 1.2.5 runing as  production system. Now it becomes
very important to my customer an exact analysis of CDRs for their QoS
to their customers.

I've been analysing the CDRs, and i notice many entries like this:

Calldate   |Channel|Source  |  Clid
 |  Dst  | Disposition | Duration
---
2006-10-24 10:10:24 | Zap/2-1... | 2023|   MSN: 2023| 100   |
FAILED  |  01:41
---

There are no complainings about dropped calls or something else.

I must say, this is a ringgroup call, and this took me into this bug:

http://bugs.digium.com/file_download.php?file_id=9084type=bug

Thanks to Mark Spencer for the attached file that appears to fix this
bug which apparently was only happening when the Dial statement
contained more than one SIP user and when those SIP users were not
connected.


Is there someone else out there using this patch on production system?
Problem Solved?

--
Best regards,

Marco Mouta
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Re: [asterisk-users] UA - number assignment

2006-10-24 Thread Marco Mouta


I think I understood what you want:
1- You want when someone dials an extension, do a Lookup in a database
using FWDCIDNAME
2- Then Dial the number that corresponds to this FWDCIDNAME in database

is that?

If it is so, i would recomend you to use AstDB - Asterisk Berkeley DB
(version1) - automatically installed with your asterisk.

Example:

exten=_X.,1,Set(NumberToDial=DB(myuserlist/${FWDCIDNAME})
exten= _X.,2,Dial(SIP/${NumberToDial})
exten= _X.,3,hangup

Take a look on this function and applications on your CLI show function DB

hope it helps.

Pls give me some feedback



On 10/24/06, Paul Ianas [EMAIL PROTECTED] wrote:





My problem is simple and I've issued it about 3 weeks ago. I want the UAs to
authenticate with a number to the SIP server. Is this possible?



For example, I configured an AT-RG613TX (Allied Telesyn Residential
Gateway). In its configuration it is not possible for me to skip specifying
a number (ex. 102) along with the username. I've looked into the source code
(SIP implementation) of Asterisk and, as I figured out, it is not possible
to tell Asterisk the number the user has.



The question is: how can I assign a number to a user in Asterisk? One
solution would be to define two rules in extensions.conf :



exten = 102,1,SetCallerId,${FWDCIDNAME}

exten = 102,2,Dial(SIP/pianas)



these would tell Asterisk that user pianas has the number 102.



Is there any other solution for my problem? (a database for example).



Thank you.



--

Paul Ianas

Programming Engineer

Level 7 Software

Timisoara, 59D Bucovinei

phone: 0744137020

email: [EMAIL PROTECTED]


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Re: [asterisk-users] Distributing calls among channels in dial group

2006-10-24 Thread Marco Mouta


Define diferent trunks for every PRI span and use RANDOM on your
dialplan before dialing!

On 10/24/06, Asterisk [EMAIL PROTECTED] wrote:

Hi everybody!

Is it possible to order Asterisk to distribute calls to ZAP channels belonging 
to one channel group (also called dial group) in any other way than in 
sequential order (1,2,3 etc.)?

I would like to distribute calls equally between all available PRI spans.

Thanks in advance for any tip!

Alex

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Com os melhores cumprimentos,

Marco Mouta
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[asterisk-users] #Transfer - Timeout is configurable?

2006-10-20 Thread Marco Mouta


Hi guys,

This should be has an easy answer for you, my users are complaining
that when they press # and then ear gorgeous Allison Transfer the
timeout is very small, they must enter immediatly the extension to
transfer the call.

Is it possible to change this?


;transferdigittimeout = 3  ; Number of seconds to wait between
digits when transfering a call

This is timeout after pressing the first digit isn't it?

--
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[asterisk-users] Orange Flash Light Mitel 5215 - Asterisk - working !

2006-10-18 Thread Marco Mouta


Hi guys,

I'm trying to reuse Mitel 5215 from proprietary system now into Asterisk :) !

I've them already with SIP and handling calls sucessfully!

I've followed instructions from:
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Mitel+5220

Additional Servers

   * Outbound Server: Off
   * Outbound Server URL: blank
   * Outbound Server Port: blank
   * Voice Mail Server: mailbox # @ * hostname or IP
   * Number of rings: 4
   * Port: 5060
   * Backup Server Timeout: 4 Seconds

This is just great, but i still with a problem! The Orange Flash light
from those phones stills turning Flashing (ON/OFF) all day. It seems
to me could be bad configuration of voicemail server parameter:
  Voice Mail Server: mailbox # @ * hostname or IP

Could you explain me what it means # here ?

As the Message button also works to retrieve voicemail messages, i
thought to put it like

[EMAIL PROTECTED]

This works good to retrieve the voicemail pressing message button, but
the Orange light keeps turning on and off all day:(


Any one can help me on this or has experience with this? Could be a
bad interpretation from me about the instructions on wiki.

Thanks,

Marco Mouta
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Re: [asterisk-users] Locking phones at night...

2006-10-18 Thread Marco Mouta


A simple way would be:

IVR menu that locks the phone, using function DB(family/key), try show
function DB
in your dialplan.

With DB(family/key) you may insert data into AstDB ; you lock the phone

you may define family= LockedPhones , and the key is the callerid
number of the phone

With DB_EXISTS(family/key) you check if there is an entry to the
current phone inside AstDB, if exists the phone is locked no calls
available else dial!

With  DBdel(family/key) you may unlock the phone from dialplan.

with asterisk -rx database get and put you can manipulate astDB from
your web application.

Hope this helps











On 10/18/06, Mojo with Horan  Company, LLC [EMAIL PROTECTED] wrote:


Tzafrir Cohen wrote:
 Is there a simple and safe way to query the astdb database outside of
 Asterisk?
after writing to it with:
asterisk -rx 'database put phones locked 1'
something like
asterisk -rx 'database get phones locked'
returns 1...

Is this what you mean by outside of asterisk?  Sorry if I misunderstood.
   I'm under the impression (possibly erroneously) that asterisk doesn't
flush its database to disk often enough for you to trust copies that
might be stored there, or to notice new changes made on-disk by you.
That being said, Berkeley DB file, /var/lib/asterisk/astdb.

Moj


--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Locking phones at night...

2006-10-17 Thread Marco Mouta


Use AstDB  to keep State from Lock and Unlock and create a simple menu
where user with a pincode may unlock his phone (also stored in Astdb,
or other database)

So  2 Menus one to lock the phone with a *XXX combination call, and
another to unlock requesting pin code.


Seems to me the simple way

On 10/17/06, Carlos Chavez [EMAIL PROTECTED] wrote:

I have a customer that wants to lock his phone when he goes home at
night so no one else can use it.  What would be the easiest way to do
this?

--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok

2006-10-16 Thread Marco Mouta

Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies
On 10/16/06, kjcsb [EMAIL PROTECTED] wrote:







The 
  problem is:Right now, and i'm referring only to calls directly handled by 
  VoiceMail application, the users get their audio files in email but the audio 
  is very very low. I've thought about changing RX gain on PRI interface 
  between legacy pbx and asterisk, but until now no complaining with audio 
  calls.


There's a patch for this: 
http://bugs.digium.com/file_download.php?file_id=10824type=bug

Cameron



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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok

2006-10-16 Thread Marco Mouta

Error syntax:is Voicemail([EMAIL PROTECTED],bg(10)) ; for busy announce and 10dB record gainOn 10/16/06, Marco Mouta 
[EMAIL PROTECTED] wrote:Thanks!But i've solved my problem only using g(#) gain argument from voicemail application! For me was enough.
Voicemail([EMAIL PROTECTED],b,g(10)) ; where 10 is the gain in dBthks guys for all your replies
On 10/16/06, kjcsb 
[EMAIL PROTECTED] wrote:








The 
  problem is:Right now, and i'm referring only to calls directly handled by 
  VoiceMail application, the users get their audio files in email but the audio 
  is very very low. I've thought about changing RX gain on PRI interface 
  between legacy pbx and asterisk, but until now no complaining with audio 
  calls.


There's a patch for this: 

http://bugs.digium.com/file_download.php?file_id=10824type=bug

Cameron



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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok

Hi Aaron!Could you please provid me your patch for 1.2? I didn't get you, it was a problem for you to get the messages into mp3 format?Did you have any problem until now with this patch on *1.2 ? My box is
1.2.5 and still very stable until now:)Hope you can help me, i can't figure out why no one though about this has a serious request on *1.2 , as this seems to happen always when you have asterisk behind a legacy pbx with zapata in telephony interface.
On 10/11/06, Aaron Daniel [EMAIL PROTECTED] wrote:
That doesn't always work :)There's two options... either port the volgain patch from 1.4 to 1.2 (Ifanyone wants a copy, we've been using it for months... however it alsoconverts to mp3 so we'd have to strip that out)... or use
1.4 whichincludes the patch.Let me know if I should post a copy of the older code somewhere.The 1.4 patch is here:http://bugs.digium.com/view.php?id=6237
Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote:
I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable.
Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution:
In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm
NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband
connection. From: Marco Mouta [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying aVoiceMailserver with Asterisk behind a legacy pbx,
providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application.
I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is:
Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and
asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience?
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[asterisk-users] Hicom 150 -- BRI -- Asterisk

Hi,Is is possible to implement this:Hicom150 --- BRI (QSIG)  AsteriskI've been reading Siemens documentation and they say:Digital nailed connectionsCorporate communication networks can be implemented over digital S0 or
S2M nailed connections between several Hicom systems using the CorNet Nprotocol and between Hicom and non-Siemens systems using the QSig protocol.The systems are linked with each other via public and/or private lines.
Does any one ever got this configuration working sucessfully?I'm wondering if it would be possible to communicate via BRI cards using QSIG.In the past i've made this successfully happened but using E1 PRI.
thanks
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Re: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok

Would you be able to tell me which lines must be reordered in app_voicemail.cOn 10/11/06, Cullin J. Wible [EMAIL PROTECTED]
 wrote:externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio
vm-audio uses 'sox -e' to determine how much to scale by without clippingand thenThen 'sox -v' to scale the sound file.This happens after the email message is sent, but by changing the order of a
few lines in the app_voicemail.c program you can have the externnotify runbefore the email message is sent.Cullin-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Aaron DanielSent: Wednesday, October 11, 2006 12:49 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Increase VoiceMail Messages Recording Gain-AudioCalls are OkThat doesn't always work :)There's two options... either port the volgain patch from 
1.4 to 1.2 (Ifanyone wants a copy, we've been using it for months... however it alsoconverts to mp3 so we'd have to strip that out)... or use 1.4 which includesthe patch.Let me know if I should post a copy of the older code somewhere.
The 1.4 patch is here:http://bugs.digium.com/view.php?id=6237Aaron DanielComputer Systems TechnicianSam Houston State University
[EMAIL PROTECTED](936) 294-4198On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the
 way up along with the speakers amps just to make the attachmentunderstandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out
 of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list.
 My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much
 difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection.  From: Marco Mouta [mailto:
[EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain -
 AudioCalls are Ok Hi all I'm deploying aVoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one
 per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no
 complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but
 the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will
 get me in troubles with Voice Calls . Any advice, or previous similar experience?___--Bandwidth and Colocation provided by 
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[asterisk-users] cdr_addon_mysql.c - Asterisk 1.4 - Asterisk Addons

Hi guys,I've been installing Asterisk 1.4 with Asterisk addons, and i could notice that in /usr/lib/asterisk/modules/ doesn't have cdr_addon_mysql.so even after compiling Asterisk Addons!In fact the cdr_addon_mysql.c exists, but it doesn't seems to be compile when i run Asterisk-Addons: make  make install
Any one can help me on this?-- best regardsMarco Mouta
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[asterisk-users] Increase VoiceMail Messages Recording Gain - Audio Calls are Ok

2006-10-10 Thread Marco Mouta

Hi allI'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions.On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application.
I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:)The problem is:Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low.
I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls.I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls .
Any advice, or previous similar experience?-- Best regardsMarco Mouta
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Re: [asterisk-users] password for vm users

2006-10-09 Thread Marco Mouta

just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password!On 10/9/06, 
stan ford [EMAIL PROTECTED] wrote:
how does one force mandatory password change on login? and a period of time to pass before mandating a password change?im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well
thx. 
		Yahoo! Messenger with Voice. 
Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.
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Re: [asterisk-users] Codes negotiation problems between Asterisk1.4beta2 and Aastra 480i

2006-10-06 Thread Marco Mouta

Have you ever tried allow=alawulaw in the same line? just a tip...On 10/6/06, Morten Isaksen [EMAIL PROTECTED]
 wrote:
On 10/6/06, Gareth Owen [EMAIL PROTECTED]
 wrote:
Morten,Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on.Can you post the INVITE message that is being rejected?



This INVITE results in a 488 from the phone:


INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK42f78e77;rportFrom: 1011 
sip:[EMAIL PROTECTED];tag=as3a35aa3aTo: 
sip:[EMAIL PROTECTED]Contact: 
sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70
Date: Fri, 06 Oct 2006 14:22:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 309
v=0o=root 4746 4746 IN IP4 192.168.10.2s=sessionc=IN IP4 
192.168.10.2t=0 0m=audio 10066 RTP/AVP 8 0 3 101a=rtpmap:8 PCMA/8000
a=ptime:20a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:3 GSM/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv
And this INVITE works (only alaw is enabled):
INVITE sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK3c04692a;rportFrom: 1011 
sip:[EMAIL PROTECTED];tag=as39cd0724To: 
sip:[EMAIL PROTECTED]Contact: 
sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70
Date: Fri, 06 Oct 2006 14:23:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 238
v=0o=root 4762 4762 IN IP4 192.168.10.2s=sessionc=IN IP4 
192.168.10.2t=0 0m=audio 10042 RTP/AVP 8 101a=rtpmap:8 PCMA/8000
a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=sendrecv
Also, I know we've fixed a number of SDP related issues in 1.4.1, so if you haven't already you might want to try the 
1.4.1 beta.Info on how to get the beta is available here:
http://groups.google.com/group/Aastra-480i-Users/browse_frm/thread/8f6f0f3419ef396d



I will try that and report back here.-- Morten Isaksen
http://www.misak.dk/blog/ 

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Re: [asterisk-users] Configuration / dialplan problem

2006-10-03 Thread Marco Mouta

If you really want _07. to be tested afterall the above patternmatches, you must define it in other context and add it as an include for the current context.Asterisk first will look for your patternmatches in the current context and oonly after this will lookup your include context. This way you can avoid the asterisk resort!
pls give some feedback if it helps...On 10/3/06, Kevin Smith [EMAIL PROTECTED] wrote:
There are a few things to look at.First off, you have a lot of wildcard testing that is probably throwing
the dial plan off. For example, you have the following:exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07.,1,Congestion()If I left it in this order what would happen? From what I understand itis nautral to think in that order, but really Asterisk is going to sortthe extensions something like this:
exten = _07.,1,Congestion()exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
So now say you dial 07545865143254/8564, it will go to the Congestionapplication every time.What I would do is comment out the wildcard searches and see if thatresolves the problem. If so, try putting all the wildcard tests in an
include and see if that helps.Take a look at these to articles as well:http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sortingAlso just out of observation, why all the testing? Seems to me you could
streamline that code down a bit more. For example, the 01 and 02 tests.If you know they are dialing N number of digits, make the test_01XX, so you know they have to dial a certain amount of digitsto be a valid call. Why send a 4 digit number out your trunk if you know
it isn't going anywhere? If you need to dial '0' then 10 digits, try this: _01NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) _02NXXNX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) _07956X,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) 3
etc.Hopefully that will help,KevinMark Muffett wrote: I have my extensions.conf set up as follows: exten = _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
 exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _01.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _02.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
 exten = _0800.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _0845.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _0870.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN}) exten = _09.,1,Congestion()
 exten = _00.,1,Congestion() exten = _07.,1,Congestion() (where nn are actually real digits). I would expect this to let me dial the 07956nn numbers etc while
 stopping dialing to other 07... numbers, but it seems to stop dialling to any 07... number including the 3 specifically listed. Any ideas? Thanks Mark ___
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Re: [asterisk-users] detecting busy on queue transfer

does it solve the problem with j option?Do you have autofallthrough=yes in your general section of extensions.conf ?autofallthrough: New in 1.2. From the sample extensions.conf:
If autofallthrough is set, then if an extension runs out of things to
do, it will terminate the call with BUSY, CONGESTION or HANGUP
depending on Asterisk's best guess (strongly recommended). If
autofallthrough is not set, then if an extension runs out of things to
do, asterisk will wait for a new extension to be dialed (this is the
original behavior of Asterisk 1.0 and earlier).
On 10/2/06, lenz [EMAIL PROTECTED] wrote:
That's my fault in the example - I forgot to add in the j. Anyway whatis strange is that I get my dialplan to jump to position 108, but at thatpoint the agent is disconnected. I thought that when falling out of the
queuetransfer context, the control would be returned to the trasferer,after hearing the I'm sorry tone. Anything I'm missing here?l.In data Mon, 02 Oct 2006 00:36:30 +0200, Marco Mouta
[EMAIL PROTECTED] ha scritto: Hi, I've been looking the application dial on my asterisk server 1.2.9, and as far
 CLI show application Dial j- Jump to priority n+101 if all of the requested channels were busy. It means that the application Dial on Asterisk 1.2 doesn't jump automatically on Busy to the extension n+101, only if you Dial it with
 j argument!exten = _0.,7,Dial(Zap/g1/${EXTEN:1},,j)exten = _0.,108,NoOp(Got busy here) Or you should handle it on you priority 8 in your dialplan
 exten = _0.,7,Dial(Zap/g1/${EXTEN:1}) exten = _0.,8,Goto(s-${DIALSTATUS},1)  This is just an example. Hope it helps, please give me some feeback.
 On 10/1/06, Lenz [EMAIL PROTECTED] wrote: I don't think that is the case - if I add a wait(10) after the step 108,
 i.e. the busy detection, the agent seems to be disconnected immediately at the dial(), not after 10 seconds. That is what made me wonder what was going on. Yours
 l. On Sun, 01 Oct 2006 17:49:15 +0200, Adam Goryachev [EMAIL PROTECTED]
 wrote:   Try adding this line:  exten = _0.,109,Hangup   Dunno if it will solve it, but might help :)   Regards,
  Adam  -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___
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Re: [asterisk-users] Siemens Hipath - asterisk, pri problem

please post your from-zaptel context in extensions.confOn 10/2/06, bivio [EMAIL PROTECTED] wrote:
Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones.
asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri.
the pri in the hipath is configured as EURO PP (with CRC4)i've already checked all previust post about this topic without any clue.many thanks/etc/zaptel.confspan=1,0,0,ccs,hdb3,crc4,yellow
bchan=1-15dchan=16bchan=17-31loadzone=itdefaultzone=it/etc/asterisk/zapata.conf[trunkgroups][channels]language=itcontext=from-zaptel

signalling=pri_netswitchtype=euroisdnrxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yes

usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=no
echotraining=800rxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nochannel = 1-15channel = 17-31---thats the debug when i try to call from the hipath
 Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [6c 05 01 80 31 30 30]

 Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ]
 [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility) -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90)
 [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 1/0x1) (Originator)
 Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
 Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 9f] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved
 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] [6c 05 01 80 31 30 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (
E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4)-- Making new call for cr 1
-- Processing Q.931 Call Setup-- Processing IE 4 (cs0, Bearer Capability)-- Processing IE 24 (cs0, Channel Identification)-- Processing IE 108 (cs0, Calling Party Number)-- Processing IE 125 (cs0, High-layer Compatibility)
 -- Extension '' in context 'from-zaptel' from '100' does not exist. Rejecting call on channel 0/31, span 1NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated Protocol Discriminator: 
Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)
 Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
--byebivio

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Re: [asterisk-users] Siemens Hipath - asterisk, pri problem

try this:first: immediate=no ; otherwise what you r saying is to asterisk automatically dial when you hook up the phone!The problem is that in your context from-zaptel you are not dialing anywhere!i couldn't find you using any Dial(...)
That's why it doesn't work! try this:[from-zaptel]exten= _X.,1,Dial(Zap/G1/${EXTEN})exten= _X.,2,hangupUse it to dial a local extension, i suppose to dial out you are using a prefix
On 10/2/06, bivio [EMAIL PROTECTED] wrote:
2006/10/2, Marco Mouta [EMAIL PROTECTED]:

please post your from-zaptel context in extensions.confThanks to Giordano (immediate=yes) i see the first improvement now i hear the asterisk voice who says the number you digited is not in use, please check
now i'm tryng to understand how to andle the from-zaptel context, here it is[from-zaptel]exten = _X.,1,Set(DID=${EXTEN})exten = _X.,n,Goto(s,1)exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})exten = s,n,NoOp(DID is now ${DID})exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1); If there's no ext-did,s,1, that means there's not a no did/no cid route. Hangup.exten = s,n,Macro(hangup)exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})exten = s,n,Macro(from-zaptel-${CHAN},${DID},1); If nothing there, then treat it as a DIDexten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1) 

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Re: [asterisk-users] Issues with calling certain phone numbers...

when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06, Luca Corti 
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbersalso via SIP through an AS5350 which has an E1 ISN PRI attached.I have a PSTN operator number (say 012345678) routed to three SIPextensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00  XX  99).[outsidetoinside]exten = 012345678,1,Dial(SIP/01,10,t);exten = 012345678,n,Dial(SIP/21,10,t);exten = 012345678,n,Dial(SIP/20,10,t);
exten = 012345678,n,Goto(1);exten = _98765432XX,1,Dial(SIP/${EXTEN:8},60);exten = _98765432XX,n,Hangup();All two digits numbers dialed from extensions are routed to otherextensions, three digit numbers get routed to the PSTN Gateway.
[insidetooutside]exten = 012345678,1,Dial(SIP/01);exten = 012345678,n,Hangup();exten = _98765432XX,1,Dial(SIP/${EXTEN:8});exten = _98765432XX,n,Hangup();exten = _XX.,1,Set(CALLERID(number)=012345678);
exten = _XX.,n,Dial(SIP/[EMAIL PROTECTED]);exten = _XX.,n,Hangup();exten = _XX,1,Dial(SIP/${EXTEN},30,t);exten = _XX,n,Hangup();The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) isnot actually routed to the PSTN gateway (as it should).I tried debugging SIP and see no request made to the AS5350. Is there acommand in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?TIALuca___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] Issues with calling certain phone numbers...

My mistake sorry for last postOn 10/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06, 
Luca Corti 
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbersalso via SIP through an AS5350 which has an E1 ISN PRI attached.I have a PSTN operator number (say 012345678) routed to three SIP
extensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00  XX  99).[outsidetoinside]exten = 012345678,1,Dial(SIP/01,10,t);exten = 012345678,n,Dial(SIP/21,10,t);exten = 012345678,n,Dial(SIP/20,10,t);
exten = 012345678,n,Goto(1);exten = _98765432XX,1,Dial(SIP/${EXTEN:8},60);exten = _98765432XX,n,Hangup();All two digits numbers dialed from extensions are routed to otherextensions, three digit numbers get routed to the PSTN Gateway.
[insidetooutside]exten = 012345678,1,Dial(SIP/01);exten = 012345678,n,Hangup();exten = _98765432XX,1,Dial(SIP/${EXTEN:8});exten = _98765432XX,n,Hangup();exten = _XX.,1,Set(CALLERID(number)=012345678);
exten = _XX.,n,Dial(SIP/[EMAIL PROTECTED]);exten = _XX.,n,Hangup();exten = _XX,1,Dial(SIP/${EXTEN},30,t);exten = _XX,n,Hangup();The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) isnot actually routed to the PSTN gateway (as it should).I tried debugging SIP and see no request made to the AS5350. Is there acommand in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?TIALuca___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] extensions.conf strangeness

2006-10-01 Thread Marco Mouta

 [invalid] exten = _X!,1,Answer() exten = _X!,2,Background(pbx-invalid)Are you sure that your invalid context is correctly written?I've never heard about this pattern match _X!As far as i know the wild card is the .
So your invalid context should be: [invalid]exten = _X.,1,Answer()
exten = _X.,2,Background(pbx-invalid)This may be the causeHope it helps.On 9/29/06, Brian Candler 
[EMAIL PROTECTED] wrote:On Fri, Sep 29, 2006 at 07:49:00PM +0200, Michael Neuhauser wrote:
 The order of include statements is important in 1.2, I don't know if this still holds for trunk/1.4. Could you please try to include the 'invalid' context as the last one (i.e., AFTER include = test, not
 before) in both internal and from-sip and then test again?Yes, this works - both contexts now behave the same.But what I don't understand is, why it worked in one context but not in theother, when both just included the same four other contexts in the same
order. Is the context merging non-deterministic? Or is it somehow sensitiveto whether the incoming call came from zaptel or SIP?Regards,Brian.___
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Re: [asterisk-users] Fax detection ...

2006-10-01 Thread Marco Mouta

Why don't you look for application NVfaxDetect ? are you using Digium boards?I've been using it sucessfully for fax reception!Look for it on voip wiki.On 10/1/06, 
Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote: Well, fax detection isn't entirely reliable anyway.Even if you assume that your fax detection feature and operation is flawless in properly
 detecting fax tones (and that most likely would be a specious assumption), not all calling fax machines send fax tones.So, y'know, that assertion gets made a lot.What's the turn rate of fax machines in the market? 3 years?5?CNG
tones are *well* over 10 years old, no?What percentage of fax calls are sent without CNG tones these days?Cheers,-- jra--Jay R. Ashworth
[EMAIL PROTECTED]DesignerBaylink RFC 2100Ashworth  AssociatesThe Things I Think'87 e24St Petersburg FL USA
http://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later,they stop having sex with you.-- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] detecting busy on queue transfer

2006-10-01 Thread Marco Mouta

Hi,I've been looking the application dial on my asterisk server 1.2.9, and as far CLI show application Dialj - Jump to priority n+101 if all of the requested channels were busy.
It means that the application Dial on Asterisk 1.2 doesn't jump automatically on Busy to the extension n+101, only if you Dial it with j argument!exten = _0.,7,Dial(Zap/g1/${EXTEN:1},,j)exten = _0.,108,NoOp(Got busy here)
Or you should handle it on you priority 8 in your dialplanexten = _0.,7,Dial(Zap/g1/${EXTEN:1})exten = _0.,8,Goto(s-${DIALSTATUS},1)This is just an example.
Hope it helps, please give me some feeback.On 10/1/06, Lenz [EMAIL PROTECTED] wrote:
I don't think that is the case - if I add a wait(10) after the step 108,i.e. the busy detection, the agent seems to be disconnected immediately at
the dial(), not after 10 seconds. That is what made me wonder what wasgoing on.Yoursl.On Sun, 01 Oct 2006 17:49:15 +0200, Adam Goryachev
[EMAIL PROTECTED] wrote: Try adding this line: exten = _0.,109,Hangup Dunno if it will solve it, but might help :) Regards, Adam
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[asterisk-users] When does Scalability requests Asterisk to Use SER ?

2006-09-19 Thread Marco Mouta

Hi all,I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions?Can i trust in a solution only with Asterisk to make all this install?
Please help me with your experience on this kind of asterisk solutions.I've googled and read about asterisk at large scale solutions, but still in doubt.
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Re: [asterisk-users] grandstream gxp 2000 does not display names when calling out

2006-09-19 Thread Marco Mouta

test it with someextension and using a sipphone like xlite or something else and you will be able to understand if it is a grandstream issue!hope it helpsOn 9/19/06, 
Christopher Corn [EMAIL PROTECTED] wrote:
i have trixbox running, the latest version and when i make an outgoing call from this phone it doesn't pick up the user's name, but instead display the number. is this a grandstream problem? or asterisk?   
  i did specify the user name from 'extension within trixbox. in matter of face, if i call into that gxp phone, it will display that users name, just doesn't do so when calling out from that phone.
any help is appreciated. thanks.
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Re: Re[2]: [asterisk-users] SHSU asterisk installation?

2006-09-16 Thread Marco Mouta

yes in did, this could be one excellent case study for all asterisk Community! Please let me know when it will be held!

By the way, keep asterisk community following your steps would be great for knowledge of everyone and to solve possible problems you could find!
On 9/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:
That would be superb!-Original Message-From: Aaron Daniel 
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comCc:Sent:Sat, 16 Sep 2006 11:32:00 -0500
Delivered:Sat,16 Sep 2006 13:21:48Subject:[asterisk-users] SHSU asterisk installation?Maybe I should just hold a conference call about all this stuff.On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Roy Sigurd Karlsbakk wrote:  Linked from /. today,  
http://www.networkworld.com/news/2006/091206-von-sam-houston.html  talks aboutthe Sam Houston State University (SHSU) migrating a  rather large amount of users to asterisk. The article describes the
  installation in rather vague terms, so I was wondering if someone  know how they plan to do this, in detail.   Yes, there have been several threads about this.
   Obviously I _have_ tried to google about this, so if you could point me  to one of those threads, I'd be grateful - From 3 days ago: 
http://www.sineapps.com/news.php?rssid=1509 Thread here: [asterisk-users] University switches to Asterisk - -- Cheers, Matt Riddell ___
 http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone)
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E-mail classificado pelo Identificador de Spam Inteligente Terra.Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15--Original Message EndsMelcon Moraes 
[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,
Marco Mouta 
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[asterisk-users] Deploying an IVR - direct extensions.conf or AGI scripts?

2006-09-12 Thread Marco Mouta

Hi all,I'm developing an IVR that will have to make some MYSQL queries and diferent DTMF menus. Preventing already my development effort, future I plan to deploy my own website where users can build their own IVR.
Would you recomend me to make it with Realtime Extensions, do it directly in extensions.conf and for queries and something else use AGI scripts, or you recomend me to build specific AGIscripts with IVR menus inside (this looks very limited for future WebConfig interface)?
What is your advice, concerning with your experience.-- Best regards,Marco Mouta
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[asterisk-users] Handling incoming calls from VoIPbuster

2006-09-11 Thread Marco Mouta

Hi all,Currently i've made some tests with VoIP buster and everything is running ok for outbound calls.Now i've created new VoIPbuster account, and my goal is to allow VoIPbuster partners to dial into my dialplan IVRs for free.
But I always get my VoIPbuster account (currently registred with my asterisk server) offline.Any one has successfull configuration for this?-- Best regards,Marco Mouta
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Re: [asterisk-users] Forward recorded voicemail message to more than one extension using sendvoicemail=yes

2006-09-11 Thread Marco Mouta

Hi,The answer for your question is Yes for Sure.Check VoiceMail( ) application syntax:VoiceMail([flags][EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]
)http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail
Hope it helps,MOn 9/11/06, Jay Dutt [EMAIL PROTECTED] wrote:
I'd like to be able to record a voicemail message, then enter a list ofextensions to forward that voicemail message. Currently, thesendvoicemail=yes setting in voicemail.conf [general] section onlyallows for one extension to forward the message. Is thee any way to
forward to multiple extensions?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

please post also your extensions.conf !;; inbound group;[inbound]; Giorgio test:pmp_l1_check=no;msns=*This Line must be uncomment for sure, of course you may match only specific msns, but for now keep it msns=*
msns=*; end testports = 1context = outbound_isdnOn 9/6/06, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:Hi Marco,in attachment you can find my 
misdn.conf. Consider that I'm still fixingsome warning because I recently upgraded from install-misdn toinstall-misdn-mqueue but the driver installation manual has not changed.Some comments are due to the fact I'm still making tests to solve the
incoming calls problem I mentioned.Thank you.Giorgio IncantalupoMarco Mouta wrote: Please post your *misdn*-*init*.*conf as well as misdn.conf so i can try to help u*
 On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote: Hi, I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23 package installed. I can make outbound calls but cannot receive any. I get no Asterisk
 messages on the console except for these: P[ 1] GOT IGNORE SETUP P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?
 TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by 
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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users[general]debug = 5tracefile = /var/log/asterisk/misdn.trace
trace_calls = falsetrace_dir = /var/log/asterisk/misdnbridging = yesstop_tone_after_first_digit = yesappend_digits2exten = yesl1_info_ok = yesclear_l3 = nomethod = standard;;; CRYPTION STUFF
dynamic_crypt = nocrypt_prefix = **crypt_keys = test,muh; users sections:[default]context = misdnlanguage = itnationalprefix = 0internationalprefix = 00rxgain = 0txgain = 0
te_choose_channel = nodialplan = 0use_callingpres = yesechocancelwhenbridged = noechotraining = yes;; inbound group;[inbound]; Giorgio test:pmp_l1_check=no;msns=*; end test
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Re: [asterisk-users] core dumps

post your tail /var/log/asterisk/fullOn 9/6/06, Anthony Musaluke [EMAIL PROTECTED] wrote:
Hell all,I have asterisk dying and restarting. After it dies, it creates a core file.Running gdb on this file produces the details below. This is very random andI am not sure what could be happening. Any ideas where I need to look?
ThanksAnthony--(gdb) bt full#00x00881402 in __kernel_vsyscall ()No symbol table info available.#10x001d38f8 in raise () from /lib/libc.so.6
No symbol table info available.#20x001d5068 in abort () from /lib/libc.so.6No symbol table info available.#30x00208a0a in __libc_message () from /lib/libc.so.6No symbol table info available.#40x0020f8ca in _int_malloc () from /lib/libc.so.6
No symbol table info available.#50x002109b9 in calloc () from /lib/libc.so.6No symbol table info available.#60x00727051 in sip_alloc (callid=0x0, sin=0x0, useglobal_nat=0,intended_method=3) at chan_sip.c:3076
p = (struct sip_pvt *) 0x20e4ad__PRETTY_FUNCTION__ = sip_alloc#70x0073100e in sip_poke_peer (peer=0x8b19a60) at chan_sip.c:11629__PRETTY_FUNCTION__ = sip_poke_peer
#80x0073dd87 in sip_poke_peer_s (data="" at chan_sip.c:5761No locals.#90x08056778 in ast_sched_runq (con=0x8b251f8) at sched.c:373x = 0res = 0#10 0x00747ddf in do_monitor (data="" at chan_sip.c:11523
res = 0sip = Variable sip is not available.--___
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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).put msns=*also test this:[outbound_isdn]exten= _X.,1,Answer()
exten = _X.,n,Playback(vm-goodbye)exten = _X.,n,Hangupexten= s,1,Answer()
exten = s,n,Playback(vm-goodbye)
exten = s,n,HangupEnable an higher debug level for misdn messages in misdn.conf (I think is this the file).Pls post your results Asterisk CLI.On 9/6/06, 
Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi Marco,I have not a normal extensions.conf:[outbound_isdn]include = parkedcallsexten = _X.,1,DeadAGI(exten2.py)exten = _X.,2,Hangupexten = s,1,DeadAGI(exten2.py)exten = s,2,Hangup
The problem is I do not see the usual coloured output on Asteriskconsole (-- Executing DeadAGI(SIP/8-1d1a, exten2.py) in new stack)even with verbose 100. It seems like extensions.conf
 is not considered.I looked for msns parameter on internet but it is not very clear what itmeans...is it for inbound or outbound calls?Thank YouGiorgio IncantalupoMarco Mouta wrote:
 please post also your extensions.conf ! ; ; inbound group ; [inbound] ; Giorgio test: pmp_l1_check=no ;msns=* This Line must be uncomment for sure, of course you may match only
 specific msns, but for now keep it msns=* msns=* ; end test ports = 1 context = outbound_isdn On 9/6/06, *Giorgio Incantalupo*  
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marco, in attachment you can find my misdn.conf
. Consider that I'm still fixing some warning because I recently upgraded from install-misdn to install-misdn-mqueue but the driver installation manual has not changed. Some comments are due to the fact I'm still making tests to solve the
 incoming calls problem I mentioned. Thank you. Giorgio Incantalupo Marco Mouta wrote:  Please post your *misdn*-*init*.*conf as well as 
misdn.conf so i can  try to help u*   On 9/5/06, *Giorgio Incantalupo* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]  mailto:[EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote:   Hi,  I hava an Asterisk box with a monoBRI + install-misdn-mqueue  0.3.1-rc23  package installed.
  I can make outbound calls but cannot receive any. I get no Asterisk  messages on the console except for these:   P[ 1] GOT IGNORE SETUP
  P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]  P[ 1] release_chan: Ch not found!   Is there anybody who can help me, please?  
  TIAGiorgio Incantalupo  ___  --Bandwidth and Colocation provided by 
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  --  Com os melhores cumprimentos,   Marco Mouta  
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http://Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users  [general] debug = 5 tracefile = /var/log/asterisk/misdn.trace trace_calls = false
 trace_dir = /var/log/asterisk/misdn bridging = yes stop_tone_after_first_digit = yes append_digits2exten = yes l1_info_ok = yes clear_l3 = no method = standard
 ;;; CRYPTION STUFF dynamic_crypt = no crypt_prefix = ** crypt_keys = test,muh ; users sections: [default] context = misdn
 language = it nationalprefix = 0 internationalprefix = 00 rxgain = 0 txgain = 0 te_choose_channel = no dialplan = 0 use_callingpres = yes
 echocancelwhenbridged = no echotraining = yes ; ; inbound group ; [inbound] ; Giorgio test: pmp_l1_check=no ;msns=*
 ; end test ports = 1 context = outbound_isdn ___ --Bandwidth and Colocation provided by 
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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Hi,Multiple Subscriber Number. This is a telephone number
  associated with an ETS 300 BRI line. Providers
  of ETS 300 often give you three MSNs with a BRI, although additional MSNs can
  be purchased. An ISDN terminal will ring (provide an alerting
  signal) only when calls are made to the MSN (or MSNs) entered in that
  terminal. If a terminal has no MSNs entered it will ring whenever
  there is a call to any of the MSN's on that BRI.You can have specific ports of your Beronet card handling specific MSNs, and then route it to diferent contexts into Asterisk, like handling two companies in one asterisk server.
Got it?Best regardsOn 9/6/06, Giorgio Incantalupo 
[EMAIL PROTECTED] wrote:Hi Marco,it seems that msns=* is necessaryto make Asterisk work correctly...I
do not why...We have another PBX with a monoBRI but have not this problem, maybe isthe different ISDN telco or the old misdn driver does notcomplainthe important is that now it works!!I have only to fix some warning (I hope for an update of the beronet
manual) and it seems all right.Thank you again for help!!Marco Mouta wrote: msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).
 put msns=* also test this: [outbound_isdn] exten= _X.,1,Answer() exten = _X.,n,Playback(vm-goodbye) exten = _X.,n,Hangup exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Enable an higher debug level for misdn messages in misdn.conf (I think is this the file). Pls post your results Asterisk CLI.
 On 9/6/06, * Giorgio Incantalupo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote: Hi Marco, I have not a normal extensions.conf: [outbound_isdn] include = parkedcalls exten = _X.,1,DeadAGI(exten2.py
) exten = _X.,2,Hangup exten = s,1,DeadAGI(exten2.py) exten = s,2,Hangup The problem is I do not see the usual coloured output on Asterisk console (-- Executing DeadAGI(SIP/8-1d1a, 
exten2.py) in new stack) even with verbose 100. It seems like extensions.conf is not considered. I looked for msns parameter on internet but it is not very clear
 what it means...is it for inbound or outbound calls? Thank You Giorgio Incantalupo Marco Mouta wrote:  please post also your 
extensions.conf !  ;  ; inbound group  ;  [inbound]  ; Giorgio test:  pmp_l1_check=no  ;msns=* 
  This Line must be uncomment for sure, of course you may match only  specific msns, but for now keep it msns=*   msns=*   ; end test
  ports = 1  context = outbound_isdnOn 9/6/06, *Giorgio Incantalupo*  [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:   Hi Marco,  in attachment you can find my misdn.conf
 . Consider that I'm still  fixing  some warning because I recently upgraded from install-misdn to  install-misdn-mqueue but the driver installation manual has not
  changed.  Some comments are due to the fact I'm still making tests to solve the  incoming calls problem I mentioned.  
  Thank you.Giorgio Incantalupo  Marco Mouta wrote:   Please post your *misdn*-*init*.*conf as well as
 misdn.conf so i can   try to help u* On 9/5/06, *Giorgio Incantalupo* 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  mailto: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]   mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto: [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote: Hi,   I hava an Asterisk box with a monoBRI +
 install-misdn-mqueue   0.3.1-rc23   package installed.   I can make outbound calls but cannot receive any. I get no  Asterisk
   messages on the console except for these: P[ 1] GOT IGNORE SETUP   P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]
   P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?   TIA
   Giorgio Incantalupo   ___   --Bandwidth and Colocation provided by 
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http://lists.digium.com/mailman/listinfo/asterisk-users   --   Com os melhores cumprimentos,
 Marco Mouta 
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   [general]  debug = 5  tracefile = /var/log/asterisk/misdn.trace  trace_calls = false
  trace_dir = /var/log/asterisk/misdn  bridging = yes  stop_tone_after_first_digit = yes  append_digits2exten = yes  l1_info_ok = yes
  clear_l3 = no  method = standard   ;;; CRYPTION STUFF   dynamic_crypt = no  crypt_prefix = **
  crypt_keys = test,muh   ; users sections:   [default]  context = misdn  language

Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Also your problem could be related with the Answer() you weren't answering the calls on your previous extensions.confPls test both configs with and without answer and reply your results.
On 9/6/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,Multiple Subscriber Number. This is a telephone number
  associated with an ETS 300 BRI line. Providers
  of ETS 300 often give you three MSNs with a BRI, although additional MSNs can
  be purchased. An ISDN terminal will ring (provide an alerting
  signal) only when calls are made to the MSN (or MSNs) entered in that
  terminal. If a terminal has no MSNs entered it will ring whenever
  there is a call to any of the MSN's on that BRI.You can have specific ports of your Beronet card handling specific MSNs, and then route it to diferent contexts into Asterisk, like handling two companies in one asterisk server.
Got it?Best regardsOn 9/6/06, Giorgio Incantalupo
 
[EMAIL PROTECTED] wrote:Hi Marco,it seems that msns=* is necessaryto make Asterisk work correctly...I
do not why...We have another PBX with a monoBRI but have not this problem, maybe isthe different ISDN telco or the old misdn driver does notcomplainthe important is that now it works!!I have only to fix some warning (I hope for an update of the beronet
manual) and it seems all right.Thank you again for help!!Marco Mouta wrote: msns is as far as i know, similar to DIDs but it includes the complete Dialed number (the number your customer has dialed to call you).
 put msns=* also test this: [outbound_isdn] exten= _X.,1,Answer() exten = _X.,n,Playback(vm-goodbye) exten = _X.,n,Hangup exten= s,1,Answer()
 exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Enable an higher debug level for misdn messages in misdn.conf (I think is this the file). Pls post your results Asterisk CLI.
 On 9/6/06, * Giorgio Incantalupo* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]
 wrote: Hi Marco, I have not a normal extensions.conf: [outbound_isdn] include = parkedcalls exten = _X.,1,DeadAGI(exten2.py

) exten = _X.,2,Hangup exten = s,1,DeadAGI(exten2.py) exten = s,2,Hangup The problem is I do not see the usual coloured output on Asterisk console (-- Executing DeadAGI(SIP/8-1d1a, 
exten2.py) in new stack) even with verbose 100. It seems like extensions.conf is not considered. I looked for msns parameter on internet but it is not very clear

 what it means...is it for inbound or outbound calls? Thank You Giorgio Incantalupo Marco Mouta wrote:  please post also your 
extensions.conf !  ;  ; inbound group  ;  [inbound]  ; Giorgio test:  pmp_l1_check=no  ;msns=* 
  This Line must be uncomment for sure, of course you may match only  specific msns, but for now keep it msns=*   msns=*   ; end test
  ports = 1  context = outbound_isdnOn 9/6/06, *Giorgio Incantalupo*  
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto:
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:   Hi Marco,
  in attachment you can find my misdn.conf
 . Consider that I'm still  fixing  some warning because I recently upgraded from install-misdn to  install-misdn-mqueue but the driver installation manual has not
  changed.  Some comments are due to the fact I'm still making tests to solve the  incoming calls problem I mentioned.  
  Thank you.Giorgio Incantalupo  Marco Mouta wrote:   Please post your *misdn*-*init*.*conf as well as
 misdn.conf so i can   try to help u* On 9/5/06, *Giorgio Incantalupo* 

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  mailto: 
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]   mailto:
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]  mailto: 
[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] wrote:  
   Hi,   I hava an Asterisk box with a monoBRI +
 install-misdn-mqueue   0.3.1-rc23   package installed.   I can make outbound calls but cannot receive any. I get no  Asterisk
   messages on the console except for these: P[ 1] GOT IGNORE SETUP   P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]

   P[ 1] release_chan: Ch not found! Is there anybody who can help me, please?   TIA
   Giorgio Incantalupo   ___   --Bandwidth and Colocation provided by 
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 Marco Mouta 

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   [general]  debug = 5  tracefile = /var/log/asterisk/misdn.trace  trace_calls = false
  trace_dir = /var/log/asterisk/misdn  bridging = yes

Re: [asterisk-users] app_rxfax Only Receives One Page

Try to increase your rxgain, and check you have echocancel disabledpls post your resultsOn 9/6/06, Steve Totaro 
[EMAIL PROTECTED] wrote:Doug Lytle wrote: Steve Totaro wrote:
 I am trying to setup a fax server and all I get is the first page of a multipage fax.The first page is perfect quality. I am not sure how to debug this.I have an HP DL320 with a quad
 Sangoma T1 board. You'll be much happier moving it over to HylaFAX and iaxmodem.Very easy to setup, low impact on the server and has error correction. Doug
I used the asterfax install script which installs iaxmodem.I will lookinto HylaFAX but I would love to find the solution to the one pageproblem while I pursue HylaFAX.___
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Re: [asterisk-users] app_rxfax Only Receives One Page

It has happened to me, with a x100p card, problems receiving fax that i've solved adjusting the gains.

Don't understand quite well why you say that...


On 9/6/06, Steve Underwood [EMAIL PROTECTED] wrote:
Marco Mouta wrote: Try to increase your rxgain, and check you have echocancel disabled
Better still, try leaving the gains alone. If the gain controls wereremoved completely from Asterisk, support issues would decreasedramatically.Steve pls post your results On 9/6/06, *Steve Totaro*  
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Doug Lytle wrote:
  Steve Totaro wrote:  I am trying to setup a fax server and all I get is the first page of  a multipage fax.The first page is perfect quality. 
  I am not sure how to debug this.I have an HP DL320 with a quad  Sangoma T1 board.   You'll be much happier moving it over to HylaFAX and iaxmodem.Very
  easy to setup, low impact on the server and has error correction.   Doug  I used the asterfax install script which installs iaxmodem.I will look
 into HylaFAX but I would love to find the solution to the one page problem while I pursue HylaFAX.___--Bandwidth and Colocation provided by 
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[asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...

Hi all,I think i'm missing something very very basic! I want my calls with DID 48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.Part of my extensions.conf where from-pstn is the context for all calls from pstn line is:
[from-pstn]include = from-pstn-custom ; create this context in extensions_custom.conf to include customizationsinclude = ext-did-custominclude = from-pstn-timecheck ; this has to be included otherwise it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]exten = _48XX,1,Answerexten = _48XX,n,SetVar(FROM_DID=${EXTEN})exten = _48XX,n,Playback(vm-goodbye)exten = _48XX,n,Hangup
[from-pstn-timecheck]exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:)exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI : -- Accepting call from '2132' to '4888' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
 -- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'pt') -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
 -- Executing Goto(Zap/1-1, s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1)
 -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack
 -- Executing PlayTones(Zap/1-1, ring) in new stack -- Executing NVFaxDetect(Zap/1-1, 8) in new stack -- Channel 0/1, span 1 got hangup requestAfter the hangup the call seems to keep executing Dialplan why?? Does this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?Hope some one can help me.-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...

I've solved the problem, but still not understanding very well why do i need it:I've inserted inside [ext-did-custom]exten=h,1,hangupWhy do i need this? this is not usually used to run something after an hangupcall?
thks!On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,I think i'm missing something very very basic! I want my calls with DID 48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.Part of my extensions.conf where from-pstn is the context for all calls from pstn line is:
[from-pstn]include = from-pstn-custom ; create this context in extensions_custom.conf to include customizationsinclude = ext-did-custominclude = from-pstn-timecheck ; this has to be included otherwise it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]exten = _48XX,1,Answerexten = _48XX,n,SetVar(FROM_DID=${EXTEN})exten = _48XX,n,Playback(vm-goodbye)exten = _48XX,n,Hangup
[from-pstn-timecheck]exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:)exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)exten = s,4,Goto(from-pstn-afthours,s,1)

Problem, look my Asterisk CLI : -- Accepting call from '2132' to '4888' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
 -- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'pt') -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
 -- Executing Goto(Zap/1-1, s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1)
 -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack
 -- Executing PlayTones(Zap/1-1, ring) in new stack -- Executing NVFaxDetect(Zap/1-1, 8) in new stack -- Channel 0/1, span 1 got hangup request
After the hangup the call seems to keep executing Dialplan why?? Does this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?Hope some one can help me.-- Com os melhores cumprimentos,Marco Mouta

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Re: [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...

Thank you Very MUCH I really appreciate your explanation, i wasn't getting it!On 9/5/06, Tony Mountifield 
[EMAIL PROTECTED] wrote:In article 
[EMAIL PROTECTED],Marco Mouta [EMAIL PROTECTED] wrote: I've solved the problem, but still not understanding very well why do i need
 it: I've inserted inside [ext-did-custom] exten=h,1,hangup Why do i need this? this is not usually used to run something after an hangupcall? thks!
Your problem is this line:exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)The pattern _. will match absolutely anything, and so when the line hangs up,
and Asterisk looks for the 'h' extension, it finds _. which matches, and doesthe goto back to 's'!!!You should never use _. as a pattern. If you want to match any NUMBER, you cando _X. to match two or more digits, and if you also want to match a single
digit you add a second line with _X as the extension.Using X ensures that the pattern won't match any of the special non-numericextensions like h, i, t and so on.Hope this helps.CheersTony
 On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote:   Hi all,   I think i'm missing something very very basic! I want my calls with DID
  48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.   Part of my extensions.conf where from-pstn is the context for all calls  from pstn line is:
   [from-pstn]  include = from-pstn-custom ; create this context in  extensions_custom.conf to include customizations  include = ext-did-custom
  include = from-pstn-timecheck; this has to be included  otherwise it overrides ext-did  exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]
  exten = _48XX,1,Answer  exten = _48XX,n,SetVar(FROM_DID=${EXTEN})  exten = _48XX,n,Playback(vm-goodbye)  exten = _48XX,n,Hangup   [from-pstn-timecheck]
  exten = _.,1,Goto(s,1) ; catch-all matching for calls that have  DID info (if a DID route hasn't matched them)  exten = s,1,GotoIf($[${IN_OVERRIDE} =  forcereghours]?from-pstn-reghours,s,1:)
  exten = s,2,GotoIf($[${IN_OVERRIDE} =  forceafthours]?from-pstn-afthours,s,1:)  exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)  exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI :   -- Accepting call from '2132' to '4888' on channel 0/1, span 1  -- Executing Answer(Zap/1-1, ) in new stack
  -- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack  -- Executing Playback(Zap/1-1, vm-goodbye) in new stack  -- Playing 'vm-goodbye' (language 'pt')
  -- Executing Hangup(Zap/1-1, ) in new stack  == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'  -- Executing Goto(Zap/1-1, s|1) in new stack
  -- Goto (from-pstn,s,1)  -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new  stack  -- Goto (from-pstn-reghours,s,1)  -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in
  new stack  -- Goto (from-pstn-reghours,s,2)  -- Executing Answer(Zap/1-1, ) in new stack  -- Executing PlayTones(Zap/1-1, ring) in new stack
  -- Executing NVFaxDetect(Zap/1-1, 8) in new stack  -- Channel 0/1, span 1 got hangup request   After the hangup the call seems to keep executing Dialplan why?? Does
  this is related with autofallback option in globals???   Why Hangup didn't exit dialplan?   Hope some one can help me.   --  Com os melhores cumprimentos,
   Marco Mouta  -- Com os melhores cumprimentos, Marco Mouta -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=-
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--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED]
 - http://tony.mountifield.org___--Bandwidth and Colocation provided by Easynews.com --
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Re: [asterisk-users] latest CentOS-asterisk-freepbx installation procedure

www.nerdvittles.comOn 9/5/06, Roland [EMAIL PROTECTED] wrote:
I've tried all those at voip.info.org but I just couldn't get it
right. and I don't have the luxury of time to try figure out how tomake it work by myself.any other very useful new guides you guys have? tnx___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] File structure question

Thanks Peter, I've also learned with your tips ;)On 9/5/06, Peter Bowyer [EMAIL PROTECTED] wrote:
On 05/09/06, Jay Moore [EMAIL PROTECTED] wrote:
 Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all.Someday, I hope, you'll find that 'simple' is a relative term.Perhaps if receiving accurate answers without biting off the hand of
the person helping you is too difficult for you, you'd be better offpaying for a support contract with some reputable organisation? Thatway you can do no work whatsoever yourself and enjoy never-endinghandholding at $150 per incident. That may suit you better.
Around peer-support lists, you tend to find an aversion to tellingpeople things they could easily look up or find out for themselves ina few keystrokes.You'll also notice that I took the trouble not only to answer your
question, but to come back and re-phrase my answer when I saw youhadn't understood my explanation. You got all that for free. Enjoy!Peter--Peter BowyerEmail: 
[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,
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[asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server

Hi all,Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server.phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer
This way also I would use ATA device as a Trunk without requiring an Asterisk server on every smalloffice and no need to buy many ATAs neither VoiP hardphones.Is this affordable or i'm missing already basic functions required for a production system?
-- Best regardsMarco Mouta
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Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages

Please post your misdn-init.conf as well as misdn.conf so i can try to help uOn 9/5/06, Giorgio Incantalupo
 [EMAIL PROTECTED] wrote:
Hi,I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23package installed.I can make outbound calls but cannot receive any. I get no Asteriskmessages on the console except for these:
P[ 1] GOT IGNORE SETUPP[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]P[ 1] release_chan: Ch not found!Is there anybody who can help me, please?TIAGiorgio Incantalupo___
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

Hi Tzafrir,I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...Thks,
On 9/2/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Sep 01, 2006 at 03:03:39PM +0100, Marco Mouta wrote: Hi all, I've just installed vim70, looking for vim syntax highlighting( for Asterisk.conf files) , 
http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting, and i notice that both: asterisk.vim and filetype.vimalready refer asterisk configurations. But unfortunately i couldn't get yet the highlight syntax working fine for
 my asterisk.conf files. Any one can help me?What happens if you run manually::set syntax=asterisk--Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755iax:[EMAIL PROTECTED]+972-50-7952406jabber:[EMAIL PROTECTED]
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[asterisk-users] Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)

Hi,I'm planning a solution to establish a connection between Main office company, and let's say 15 small offices:Plan:1- Asterisk Server @ Main Office (connected to main office legacy pbx)2- Linksys SPA 3000 to install on every small office.
Idea behind SPA 3000, the main goal is to keep every user with their traditional phone and just connect SPA3000 to the legacy pbx of the small office and then route the calls from lecagy PBX to MainOffice Asterisk Server via voIP.
Then SPA3000 would be used as a low cost solution to allow any user from the small office to call Main office Company. This way with only one or two ATA per small office i would be able to connected every one with main office with very lowcost price
I would like to hear from you any suggestions or ideas, is this acceptable for a productions system?-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] Any Hardphone with VPNClient embedded?

Hi all,Does any of you knows an Hardphone with VPN client embedded? -- Best regards,Marco Mouta
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

Just Great!What was missing is:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic [EMAIL PROTECTED] wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...
Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

BTW Could you tell me how to i make it load this option by default everytime?On 9/4/06, Marco Mouta 
[EMAIL PROTECTED] wrote:Just Great!What was missing is
:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic 
[EMAIL PROTECTED] wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...
Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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Re: [asterisk-users] vim syntax highlighting( for Asterisk.conf files)

Done,I've created ~/.vimrc file and inside this file:syntax onthks once moreOn 9/4/06, Marco Mouta 
[EMAIL PROTECTED] wrote:BTW Could you tell me how to i make it load this option by default everytime?
On 9/4/06, Marco Mouta 

[EMAIL PROTECTED] wrote:Just Great!What was missing is

:syntax onNow perfect! Thks guys! In fact i couldn't find this basic step any where except here. Ok I'm a newbie, but it will help others if is written in the tutorials. I'll look for wiki to post this there.
On 9/4/06, Victor Toofic 

[EMAIL PROTECTED] wrote:
On mon, sep 04, 2006, 10:44 +0100, Marco Mouta wrote: I've made vim /etc/asterisk/extensions_custom.conf then :set syntax=asterisk, and nothing happens. No errors no warnings and also no highlight syntax...
Hi!!I have asterisk.vim under /usr/share/vim/vimXX/syntax/ and:set filetype=asterisk:syntax on (optionally)works fine for me.--Víctor Toofic___
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Re: [asterisk-users] FAX handling

Pls Post your Asterisk CLI when Fax is incoming.On 9/4/06, [EMAIL PROTECTED] 
[EMAIL PROTECTED] wrote:I read on this list not so long ago that you should only enable alaw.I've
never tested this.Phil. Jose Limeres [EMAIL PROTECTED] omTo
 Sent by:Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED]
 asterisk-users@lists.digium.com m.comcc Subje
 ct
 04/09/2006 15:35[asterisk-users] FAX handling Please respond toAsterisk UsersMailing List -Non-CommercialDiscussion
 [EMAIL PROTECTED] ists.digium.comHi all,I am using asterisk 1.2.10 BRI stuffed 0.3.0-PRE-1s with zaptel 1.2.8
 andwe are trying to have FAX receiving working inone of the BRI lines.No problem with FAX transmissions but we can not receive. I have configuredin zapata.conf faxdetect=both (tx and rx).FAX machine is connected to one FXS port on a PAP2 with G711a and no echo
cancelation configured. When the FAX arrives at theFAX machine, they start negotiating but then it stops as if the format isnot recognized by the Fax machine as a valid fax.Does anyone have a similar configuration working?
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Re: [asterisk-users] File structure question

So the #include could be made just after the [general] section o extensions.conf? outside of any specific context, i think this was the question.On 9/4/06, 
Peter Bowyer [EMAIL PROTECTED] wrote:
On 04/09/06, Jay Moore [EMAIL PROTECTED] wrote: Right, I guess I was wondering if it's possible to include a file without it being in a context.The goal I wanted to achieve was to have
 as few contexts in the main extensions.conf file as possible.Did you try it? It would take... perhaps 30 seconds? A minute ifyou're a slow typist...Yes, you can do this. #include is a literal text include, as the last
poster said.--Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] help me!!Problem on incoming calls

,Playback(/tmp/asterisk-recording);exten = 
5678,5,Wait(2);exten = 5678,6,Hangup 


4) This is [from-internal-trixbox]section in 
extensions_trixbox.conffile:

[from-internal-trixbox]include = custom-speed-dial

exten = _**.,1,Pickup(${EXTEN:2}) ; GXP-2000 phone press BLF to pick up 
ringing call

exten = *61,1,Answerexten = *61,2,AGI(weather.agi)exten 
= *61,3,Hangup

exten = *62,1,Answer exten = *62,2,AGI(wakeup.php) exten 
= *62,3,Hangup 

exten = 611,1,Answerexten = 611,2,Wait(1)exten = 
611,3,DigitTimeout(7)exten = 611,4,ResponseTimeout(10)exten = 
611,5,Flite(At the beep enter the three character airport code for the weather 
report you wish to retrieve.)exten = 611,6,Read(APCODE,beep,3)exten 
= 611,7,Flite(Please hold a moment while we contact the National Weather 
Service for your report.)exten = 
611,8,AGI(nv-weather.php|${APCODE})exten = 611,9,NoOp(Wave file: 
${TMPWAVE})exten = 611,10,Playback(${TMPWAVE})exten = 
611,11,Hangup

; CallingCard application;add an incoimf route for the DID to Custom 
App: (un-comment next line);custom-callingcard,s,1

; un-comment the 6 lines below to work on incoming 
DIDs;[custom-callingcard];exten = s,1,Answer;exten = 
s,2,Wait,2;exten = s,3,DeadAGI,a2billing.php;exten = 
s,4,Wait,2;exten = s,5,Hangup

  - Original Message - 
  
From: 
  Marco 
  Mouta 
  To: 
Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, August 31, 2006 6:07 
  PM
  Subject: Re: [asterisk-users] help 
  me!!Problem on incoming calls
  forgot to mention, it may help if you post your 
  extensions.confAs you are using from-internal context for this 
  calls,and you are using trixbox, have look in extensions_additional.conf and 
  all extension_*.conf to find out your [from-internal] context. By the 
  way I wouldn't use the from-internal context for incoming calls from PSTN 
  line...
  On 8/31/06, Marco 
  Mouta  
  [EMAIL PROTECTED] wrote:
  
Hi Please Post you Asterisk CLi when incoming is arriving. 

On 8/31/06, Patrick  
[EMAIL PROTECTED]  wrote:
On 
  Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi, 
   Please Help me!!! I've installed 
  TrixBoxand VISDN (snapshot 20060802) on a PC with an 
  HFC-4s card. Outbound Calls work fine, and inbound calls from 
  Cellphones work fine too.  I have a problem with 
  incoming calls beginning with 0 (national and international 
  calls-I stay in Italy) Thanks in advance for 
  answers Andrea De NadaiWouldn't it be a better idea to 
  ask on the vISDN mailing list and/or the Trixbox forum? You are after 
  all using their 
  software...Regards,Patrick___--Bandwidth 
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  options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

-- Com os melhores cumprimentos,
Marco Mouta -- Com os melhores cumprimentos,Marco Mouta 
  
  

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Re: [asterisk-users] help me!!Problem on incoming calls

add this to [justtotest]exten=s,1,Answerexten=s,2,Playback(vm-goodbye)exten=s,3,hangupreply your results and asterisk cli
On 9/1/06, Andrea infoteam [EMAIL PROTECTED] wrote:








Hi Marco,
my from-trunk context in extensions.conf 
is:

[from-trunk]
include = from-pstn

[from-pstn]include = 
from-pstn-custom 
; create this context in extensions_custom.conf to include 
customizationsinclude = ext-findmefollow; MODIFICATOIN (PL) 
for findmefollow if enabled, should be bofore ext-localinclude = 
ext-did-direct; MODIFICATOIN (PL) put before ext-did to take 
precedenceinclude = ext-didexten = 
fax,1,Goto(ext-fax,in_fax,1)

;end from-trunk

I tested your example:
The input calls from cellphones execute the [justtotest] section,
the other input 
calls not worked fine.

  
- Original Message - 
  
From: 
  Marco 
  Mouta 
  To: 
Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, September 01, 2006 12:14 
  PM
  Subject: Re: [asterisk-users] help 
  me!!Problem on incoming calls
  Hi,Please Post your from-trunk context or clever than 
  that create a context in 
  extensions_custom.conf:[justtotest]exten=_X.,1,Answerexten=_X.,2,Playback(vm-goodbye)exten=_X.,3,hangup 
  and in your 
  [VISDN1.2]context=justtotest .Be aware 
  to write only in extensions_custom.conf , the other files are changed by 
  trixbox on updates(at least this is the beavior of freepbx from hold 
  [EMAIL PROTECTED] the previous of Trixbox.Please post your 
  results.Best regards
  On 9/1/06, Andrea 
  infoteam  
  [EMAIL PROTECTED] wrote:
  



ThanksMarco,


this is my CLI when received a call from number 
beginning wwith 0:

-- Executing Macro(VISDN/visdn1.2/76.I, hangupcall) in new 
stack -- Executing ResetCDR(VISDN/visdn1.2/76.I, 
w) in new stack -- Executing 
NoCDR(VISDN/visdn1.2/76.I, ) in new stack -- 
Executing Wait(VISDN/visdn1.2/76.I, 5) in new 
stack -- Executing Hangup(VISDN/visdn1.2/76.I, ) 
in new stack

1) this is [from-internal] and [from-pstn] 
section of my extensions.conf:

[from-internal]; applications are now mostly all found in 
from-internal-additional in _custom.confinclude = 
parkedcallsinclude = from-internal-custom;allow phones to dial 
other extensionsinclude = ext-faxinclude = 
from-trunk;allow phones to access generated contexts;; MODIFIED 
(PL);; Currently the include for findmefollow is being 
auto-generated before ext-local which is the desired behavior.; However, 
I haven't been able to do anything that I know of to force this. We need to 
determine if it should; be hardcoded into here to make sure it doesn't 
change with some configuration. For now I will leave it out; until we 
can discuss this.;include = from-internal-additionalexten 
= s,1,Macro(hangupcall)exten = h,1,Macro(hangupcall)

[from-pstn]include = 
from-pstn-custom 
; create this context in extensions_custom.conf to include 
customizationsinclude = ext-findmefollow; MODIFICATOIN 
(PL) for findmefollow if enabled, should be bofore ext-localinclude 
= ext-did-direct; MODIFICATOIN (PL) put before ext-did to 
take precedenceinclude = ext-didexten = 
fax,1,Goto(ext-fax,in_fax,1)

2) This is a [from-internal-additional] section in 
extensions_additional.conf file:

[from-internal-additional]include = 
from-internal-additional-custominclude = ext-groupinclude = 
app-dnd-offinclude = app-dnd-oninclude = 
ext-queuesinclude = app-callwaiting-cwoffinclude = 
app-callwaiting-cwoninclude = ext-meetmeinclude = 
app-cf-busy-offinclude = app-cf-busy-off-anyinclude = 
app-cf-busy-oninclude = app-cf-offinclude = 
app-cf-off-anyinclude = app-cf-oninclude = 
app-cf-unavailable-offinclude = app-cf-unavailable-oninclude 
= app-recordingsinclude = app-calltraceinclude = 
app-directoryinclude = app-echo-testinclude = 
app-speakextennuminclude = app-speakingclockinclude = 
ext-findmefollowinclude = app-gabcastinclude = 
app-dialvminclude = app-vmmaininclude = 
app-userlogonoffinclude = app-zapbargeinclude = 
ext-testinclude = ext-localinclude = 
outbound-allroutesexten = h,1,Hangup

; end of [from-internal-additional]

3) This is all extensions_custom.conf file:

#include extensions_trixbox.conf#include extensions_hud.conf

[from-restricted];; These are all the applications that you 
will require; include = app-cf-busy-offinclude = 
app-cf-busy-off-anyinclude = app-cf-busy-oninclude = 
app-cf-offinclude = app-cf-off-anyinclude = 
app-cf-oninclude = app-cf-unavailable-offinclude = 
app-cf-unavailable-oninclude = app-calltraceinclude = 
app-callwaiting-cwoffinclude = app-callwaiting-cwoninclude = 
app-dialvminclude = app-directoryinclude = 
app-dnd-offinclude = app-dnd-oninclude = 
app-echo-testinclude = app

[asterisk-users] vim syntax highlighting( for Asterisk.conf files)

Hi all,I've just installed vim70, looking for vim syntax highlighting( for Asterisk.conf files) , http://voip-info.org/tiki-index.php?page=vim+syntax+highlighting
, and i notice that both: asterisk.vim and filetype.vim
already refer asterisk configurations.But unfortunately i couldn't get yet the highlight syntax working fine for my asterisk.conf files.Any one can help me?Centos4.2 is my distribuition
-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] help me!!Problem on incoming calls

HiCalls from Cellphones and from landline phones arriving into your asterisk using the same PSTN provider? Or do you have GSM gateway?It seems to me your problem is you are not receiving any DID or MSNs when call comes from landline (i mean start with 0)
M.On 9/1/06, Andrea infoteam [EMAIL PROTECTED] wrote:







Hi marco,

this is my cli when i receive a call beginning with 0,i have done two 
tests:

First test the cli is:

 -- Executing Answer(VISDN/visdn1.2/10.I, ) in new 
stack -- Executing Playback(VISDN/visdn1.2/10.I, 
vm-goodbye) in new stack -- Playing 'vm-goodbye' 
(language 'it') -- Executing Hangup(VISDN/visdn1.2/10.I, 
) in new stack

Second test the cli is:

 -- Executing NoOp(VISDN/visdn1.1/11.I, No DID or CID 
Match) in new stack -- Executing 
Answer(VISDN/visdn1.1/11.I, ) in new stack -- 
Executing Wait(VISDN/visdn1.1/11.I, 2) in new stack -- 
Executing Playback(VISDN/visdn1.1/11.I, ss-noservice) in new 
stack -- Playing 'ss-noservice' (language 
'it') -- Executing SayAlpha(VISDN/visdn1.1/11.I, ) in 
new stack
Please,what's mean the network_specific_prefix = , 
subscriber_prefix = , abbreviated_prefix =  in section [global] of the 
visdn.conf file?

Another questions for you...you speak Italian 
language?

Thanks in advance for answers





  
- Original Message - 
  
From: 
  Marco 
  Mouta 
  To: 
Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Friday, September 01, 2006 3:23 
  PM
  Subject: Re: [asterisk-users] help 
  me!!Problem on incoming calls
  add this to [justtotest]exten=s,1,Answerexten=s,2,Playback(vm-goodbye)exten=s,3,hangupreply 
  your results and asterisk cli
  On 9/1/06, Andrea 
  infoteam [EMAIL PROTECTED] wrote:
  



 
Hi Marco,
my from-trunk context in extensions.conf 
is:

[from-trunk]
include = from-pstn


[from-pstn]include = 
from-pstn-custom 
; create this context in extensions_custom.conf to include 
customizationsinclude = ext-findmefollow; MODIFICATOIN 
(PL) for findmefollow if enabled, should be bofore ext-localinclude 
= ext-did-direct; MODIFICATOIN (PL) put before ext-did to 
take precedenceinclude = ext-didexten = 
fax,1,Goto(ext-fax,in_fax,1)


;end from-trunk

I tested your example:
The input calls from cellphones execute the 
[justtotest] 
section,
the other 
input calls not worked fine.



- 
Original Message - 

From: 
Marco Mouta 
To: 
Asterisk Users 
Mailing List - Non-Commercial Discussion 

Sent: 
Friday, September 01, 2006 12:14 PM
Subject: 
Re: [asterisk-users] help me!!Problem on incoming calls
Hi,Please Post your from-trunk context or clever than 
that create a context in 
extensions_custom.conf:[justtotest]exten=_X.,1,Answerexten=_X.,2,Playback(vm-goodbye)exten=_X.,3,hangup 
and in your 
[VISDN1.2]context=justtotest .Be 
aware to write only in extensions_custom.conf , the other files are changed 
by trixbox on updates(at least this is the beavior of freepbx from hold 
[EMAIL PROTECTED] the previous of Trixbox.Please post your 
results.Best regards


  
  
  
  
  
  
  
- 
  Original Message - 
  
From: 
  Marco Mouta 
  To: 
  Asterisk Users 
  Mailing List - Non-Commercial Discussion 
  Sent: 
  Thursday, August 31, 2006 6:07 PM
  Subject: 
  Re: [asterisk-users] help me!!Problem on incoming calls
  forgot to mention, it may help if you post your 
  extensions.confAs you are using from-internal context for this 
  calls,and you are using trixbox, have look in extensions_additional.conf 
  and all extension_*.conf to find out your [from-internal] context. 
  By the way I wouldn't use the from-internal context for incoming 
  calls from PSTN line...
  On 8/31/06, Marco 
  Mouta  
  [EMAIL PROTECTED] wrote: 
  
Hi Please Post you Asterisk CLi when incoming is arriving. 

On 8/31/06, Patrick  
[EMAIL PROTECTED]  wrote: 
On 
  Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi, 
   Please Help me!!! I've installed 
  TrixBoxand VISDN (snapshot 20060802) on a PC with 
  an HFC-4s card. Outbound Calls work fine, and inbound calls 
  from Cellphones work fine too.  I have 
  a problem with incoming calls beginning with 0 (national and 
  international calls-I stay in Italy) Thanks in advance 
  for answers Andrea De NadaiWouldn't it be a better 
  idea to ask on the vISDN mailing list and/or the Trixbox forum? 
  You are after all using their 
  software...Regards,Patrick___--Bandwidth 
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Re: [asterisk-users] Probelm with incoming calls to my DID-Please help me

Hi,Please read bellow:On 9/1/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends, Thank you to all for your response and cooperation to me. I have a doubt.We have registered with Teliax and got DID number. We are making calls to USA successfully using your service. But, We are unable to receive incoming calls to our DID. Here I am sending my config files and error message on Asterisk console. 
Contents in IAX.CONF file:disallow=all   allow = ulaw [general]  register = 
teliaxusername:[EMAIL PROTECTED]  [teliax]  context=telincoming  type=friend  host=
voip-co1.teliax.com  auth=md5  secret=teliaxpassword  disallow=all  allow=ulaw  allow=alaw  allow=gsm 
  Contents in Sip.conf file:  [105] type=friend username=105 secret=ravi callerid=RaviKanth host=dynamic
 context=leader canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED]  [107] type=friend username=107 secret=suresh callerid=Suresh host=dynamic
 context=administration canreinvite=no nat=yes dtmfmode=rfc2833 allow=all [EMAIL PROTECTED] 
 Contents in Extensions.conf file:[telincoming]  exten = 303xxx, 1, Answer()  exten = 303xxx, n, Wait,2  exten = 303xxx, n, Goto(incoming,s,1)
You need to inser _ before a pattern so asterisk can try to match it:  exten = _303xxx, 1, Answer()exten = _303xxx, n, Wait,2  exten = _303xxx, n, Goto(incoming,s,1)
Should solve your problem!Also only as debug you can try _X. Pls tell me if it solved your problem.
 include = internal  include = incoming   [incoming] 
 exten = s,1,Wait(3)  exten = s,n,Answer  exten = s,n,SetMusicOnHold(default)  exten = s,n,Set(TIMEOUT(digit)=5)  exten = s,n,Set(TIMEOUT(response)=10)  exten = s,n,Background(/tmp/virg2)
  exten = s,n,Goto(s,1)  exten = s,n,Hangup()  include = internal   [internal]  exten = 105,1,SetMusicOnHold(default)  exten = 105,2,Dial(SIP/105,7,t,m,T)  exten = 1605,1,VoiceMailMain(
[EMAIL PROTECTED])  exten = 105,3,VoiceMail([EMAIL PROTECTED])  exten = 105,4,Hangupexten = 107,1,SetMusicOnHold(default)  exten = 107,2,Dial(SIP/107,7,t,m,T)  exten = 1607,1,VoiceMailMain(
[EMAIL PROTECTED])  exten = 107,3,VoiceMail([EMAIL PROTECTED])  exten = 107,4,Hangup[uscall]  exten = _1XX,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)  [manager] include = uscall
   include = internalThe error message on Asterisk console:   *CLI -- Executing Dial(SIP/105-007951e0, IAX2/[EMAIL PROTECTED]
/1303xxx|30|tr) in new stack-- Called [EMAIL PROTECTED]/1303xxx-- Call accepted by 
207.174.202.2 (format ulaw)-- Format for call is ulaw-- IAX2/teliax-1 is ringing-- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0-- IAX2/teliax-1 is ringing
-- IAX2/teliax-1 is busy-- Hungup 'IAX2/teliax-1'== Everyone is busy/congested at this time (1:1/0/0)== Auto fallthrough, channel 'SIP/105-007951e0' status is 'BUSY'What is the
 problem? Can you please tell me the solution. Looking forward to your response. Thank you.  Regards, Chandra.  
	

	
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Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060

which softphone?On 9/1/06, Elpidio Ramos [EMAIL PROTECTED] wrote:
Does anyone knows what could be the cause for asterisk not listening in post 5060 if SIP interfaces is loaded with no problems?I am using Fedora Core 3.I have followed the instructions in several tutorials and tried several soft phones and the SIP interface seem to be dead.
1. When loading asterisk SIP load with no problem  2. When I activate the DEBUG for a peer, ip or sip in general, I don't get to see any messages when a connection is attempted from any soft phone.
  3. The soft phones all report a timeout when trying to register.  4. Tried also to move to port 80 that I know is open but still the same problem.I will appreciate any help anyone can provide with this problem.
Elpidio
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[asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!

Hi all,I'm developing a Click to call Website, but now i'm getting worried with Click to Call fraud Imagine I just create one of this PhoneNumbers (extra charged numbers: like games, erotic lines...) in a remote country
Then i just go to a click to call website and start an attack inserting this Special Phone Number like 1$/min. Even if i control call duration on asterisk, and also my recepcionist will notice this is a fraud call, i'll be charged for the 1st minute every time
My Idea would be if someone has already worried with this, wouldn't be great to create a list on wiki or something where we can share this pattern Numbers?Is very hard to discover all the patterns for all the countries without sharing our knowledge...
Any tips?-- Best regards,Marco Mouta
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Re: [asterisk-users] help me!!Problem on incoming calls

Hi Please Post you Asterisk CLi when incoming is arriving.On 8/31/06, Patrick [EMAIL PROTECTED]
 wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi,
 Please Help me!!! I've installed TrixBoxand VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and inbound calls from Cellphones work fine too.
 I have a problem with incoming calls beginning with 0 (national and international calls-I stay in Italy) Thanks in advance for answers Andrea De NadaiWouldn't it be a better idea to ask on the vISDN mailing list and/or the
Trixbox forum? You are after all using their software...Regards,Patrick___--Bandwidth and Colocation provided by Easynews.com
 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] help me!!Problem on incoming calls

forgot to mention, it may help if you post your extensions.confAs you are using from-internal context for this calls,and you are using trixbox, have look in extensions_additional.conf and all extension_*.conf to find out your [from-internal] context.
By the way I wouldn't use the from-internal context for incoming calls from PSTN line...On 8/31/06, Marco Mouta 
[EMAIL PROTECTED] wrote:Hi Please Post you Asterisk CLi when incoming is arriving.
On 8/31/06, Patrick 
[EMAIL PROTECTED]
 wrote:On Thu, 2006-08-31 at 17:27 +0200, Andrea infoteam wrote: Hi,

 Please Help me!!! I've installed TrixBoxand VISDN (snapshot 20060802) on a PC with an HFC-4s card. Outbound Calls work fine, and inbound calls from Cellphones work fine too.
 I have a problem with incoming calls beginning with 0 (national and international calls-I stay in Italy) Thanks in advance for answers Andrea De NadaiWouldn't it be a better idea to ask on the vISDN mailing list and/or the
Trixbox forum? You are after all using their software...Regards,Patrick___--Bandwidth and Colocation provided by 
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 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users
-- Com os melhores cumprimentos,Marco Mouta

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Re: [asterisk-users] Asterisk Sending Data to a Web Page

Hi,As far as I know you must have a look on Asterisk Manager Interface, the HTTP way to communicate with asterisk and send and receive commands/call states etcHave a look on wiki for AMI, or Asterisk Manager Interface.
On 8/31/06, David R. [EMAIL PROTECTED] wrote:
How do I get Asterisk to send streaming data, such as incoming calls, call times, etc. to a web page? I have a web app that I'm trying to use as a call manager.Thanks,David


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Re: [asterisk-users] Help Preventing Click to Call fraud on Asterisk Servers!