Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

[Charles Wang]

Hi all,

I found the answer about it.
First, I must turn off callwaiting & callwaitingcallerid from
chan_dahdi.conf.
Second, I can't add tTkK parameters after dial(related with DTMF).
Third, I can't add DYNAMIC_FEATURES before dial.

By this way, I can get Native Bridge.

Best regards,
Charles

2015-01-30 9:16 GMT+08:00 Charles Wang :

> Hi Richard,
>
> Thank you for your response. But after I remove the parameters of dial
> command (tTkK). The call was still not native bridge.
> Let me know if you have any suggestion.
>
> Best regards,
> Charles
>
> 2015-01-30 0:34 GMT+08:00 Richard Mudgett :
>
>>
>>
>> On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang 
>> wrote:
>>
>>> Hi all,
>>>
>>> I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk
>>> 11.14.2 and DAHDI 2.8.0.
>>>
>>> I try to set callwaiting = no AND callwaitingcallerid = no in
>>> chan_dahdi.conf.
>>> But I can't find native bridging information from CLI(opened debug mode
>>> in logger.conf). How can I test the dahdi_bridge in native bridge mode?
>>>
>>> I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to
>>> FXS2.
>>>
>>> Does anyone kind to help me solve it?
>>>
>>
>> Native bridging cannot happen if Asterisk has an interest in the audio
>> stream.
>> Remove the tTkK flags in the Dial command.
>>
>> Richard
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Regards
> Charles
>



-- 
Best Regards
Charles
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Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

[Charles Wang]

Hi Richard,

Thank you for your response. But after I remove the parameters of dial
command (tTkK). The call was still not native bridge.
Let me know if you have any suggestion.

Best regards,
Charles

2015-01-30 0:34 GMT+08:00 Richard Mudgett :

>
>
> On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang 
> wrote:
>
>> Hi all,
>>
>> I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk
>> 11.14.2 and DAHDI 2.8.0.
>>
>> I try to set callwaiting = no AND callwaitingcallerid = no in
>> chan_dahdi.conf.
>> But I can't find native bridging information from CLI(opened debug mode
>> in logger.conf). How can I test the dahdi_bridge in native bridge mode?
>>
>> I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to
>> FXS2.
>>
>> Does anyone kind to help me solve it?
>>
>
> Native bridging cannot happen if Asterisk has an interest in the audio
> stream.
> Remove the tTkK flags in the Dial command.
>
> Richard
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Charles
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[asterisk-users] What conditions allow the use of dahdi native bridge?

[Charles Wang]

Hi all,

I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk
11.14.2 and DAHDI 2.8.0.

I try to set callwaiting = no AND callwaitingcallerid = no in
chan_dahdi.conf.
But I can't find native bridging information from CLI(opened debug mode in
logger.conf). How can I test the dahdi_bridge in native bridge mode?

I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to
FXS2.

Does anyone kind to help me solve it?

-- 
Best Regards
Charles
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Re: [asterisk-users] Delaying retry since we're currently running

[Charles Wang]

Hi, all

I also got the same trouble.
If the permission of call file was wrong, asterisk should not add lines
after the tail of call files such as DelayRetry .

Does anyone help me to solve it?

My call file is:
=
Channel:SIP/192.168.1.200/01124
Callerid:
MaxRetries:0
RetryTime:600
WaitTime:60
Context:from-1
Extension:01124
Priority:1

StartRetry: 3284 1 (1391598647)

DelayedRetry: 3284 0 (1391598646)

DelayedRetry: 3284 0 (1391598647)

DelayedRetry: 3284 0 (1391598647)
(many the same delayretry information skips)


Best regards,
Charles


2012-12-28 Danny Nicholas :

> My best guess is that you are creating the .call file with permissions
> that don’t allow Asterisk to delete it when it is finished or retries have
> been exhausted.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir
> *Sent:* Friday, December 28, 2012 7:49 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Delaying retry since we're currently running
>
>
>
> Hi,
>
>
>
> I am making 200 call concurrently via call files. But i get these messages
> in asterisk logs:
>
>
>
> *Delaying retry since we're currently running*
>
>
>
>
>
> Also, in call files i have  the following lines:
>
>
>
> *DelayedRetry: 28662 0 (1356701828)*
>
> *DelayedRetry: 28662 0 (1356702128)*
>
> *DelayedRetry: 28662 0 (1356702428)*
>
>
>
>
>
> I set MaxRetries: 0. I did not understand the problem, any idea?
>
>
>
>
>
> --
> Necati DEMİR
> 
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Charles
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[asterisk-users] Does cdr adaptive odbc re-connect automatically after a long idle time?

[Charles Wang]

Hi all,

I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc
to write CDR to my MySQL's cdr table.
After my testing, this scenario is working well.

After a long idle time, I didn't make any call to the asterisk server.
When I try to make a call again after 8 hours, I found that the cdr lost.
It cannot be inserted to cdr table.
Also, I could not find the insert CDR messages in the CLI at this period.

Could you please tell me which settings are wrong? Why dose my odbc
connection not re-connect to MySQL automatically?


I checked the setting below:

CLI:
ubuntu*CLI> cdr show status

Call Detail Record (CDR) settings
--
  Logging:Enabled
  Mode:   Simple
  Log unanswered calls:   Yes
  Log congestion: Yes

* Registered Backends
  ---
cdr-custom
Adaptive ODBC
csv

ubuntu*CLI> odbc show all

ODBC DSN Settings
-

  Name:   asterisk
  DSN:asterisk-connector
Last connection attempt: 2014-01-11 18:16:40
  Pooled: Yes
  Limit:  1000
  Connections in use: 0


-- /etc/asterisk/cdr.conf lists below:
[general]
enable=yes
unanswered = yes
congestion = yes
endbeforehexten=yes

[csv]
usegmtime=no; log date/time in GMT.  Default is "no"
loguniqueid=yes  ; log uniqueid.  Default is "no"
loguserfield=yes ; log user field.  Default is "no"
accountlogs=yes  ; create separate log file for each account code. Default
is "yes"

-- /etc/odbc.ini
[asterisk-connector]
Description   = MySQL connection to 'asterisk' database
Driver= MySQL
Database  = mydatabase
Server= localhost
UserName  = root
Password  = mypassword
Port  = 3306
Socket= /var/run/mysqld/mysqld.sock


-- /etc/asterisk/res_odbc.conf lists below:
[ENV]

[asterisk]
enabled => yes
dsn => asterisk-connector
password => mypassword
pre-connect => yes
sanitysql => select 1
pooling => yes
idlecheck => 30
share_connections => yes
limit => 1000
connect_timeout => 60
negative_connection_cache => 600


-- /etc/asterisk/cdr_adaptive_odbc.conf lists below:
[cdr]
connection=asterisk
table=cdr
alias start => calldate
alias phoneno => phoneno
alias userid => userid
alias callerid => callerid


-- 
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Charles
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[asterisk-users] (CALL FILES to Local Channel)billsec Zero in cdr via cdr_adaptive_odbc

[Charles Wang]

Hi, all

Sorry for null subject last mail.

I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it "Asterisk11".
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then Asterisk11 will generate a CDR records to MySQL's cdr table(in
database "mydatabase") via cdr_adaptive_odbc.
The "SIP/A221" is another asterisk machine named it "Elastix24".

I have two BIG QUESTIONs about cdr_adaptive_odbc.

First, I have answered call from Elastix24 and I can listen the music file
played from Asterisk11.
In another word, this call should be answered and its billsec is greater
than 0.

Second, if I don't want to use forkcdr(), how to config it and I can get
another cdr record that call from SIP/A221(Elastix24) to my Exten:77?

I know that the outgoing file will make a call to Local Channel and try to
Dial SIP/A221.
If it answered, this old channel should be hangup and generate another new
channel to connect to Extension:77(my callback exten).

I can't find two cdr records in mycdr table.
mysql> select * from gvl_cdr;
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid | src | dst   | dcontext| channel
  | dstchannel| lastapp |
lastdata | duration | billsec | disposition | amaflags |
accountcode | userfield | uniqueid | linkedid | sequence |
peeraccount | phoneno | callerid | userid |
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:37:01 |  | |77 | from-internal-out-7 |
Local/77@from-internal-out-7-;2   | SIP/A221- |
Dial| SIP/A221/77,30   |   17 |   0 | ANSWERED|
   3 | |   | 1389163021.1 | 1389163021.0 | 1|
  | 77  |  |  7 |



Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the
3th one).
mysql> select * from gvl_cdr;
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid   | src | dst   |
dcontext| channel|
dstchannel| lastapp | lastdata  | duration |
billsec | disposition | amaflags | accountcode | userfield | uniqueid |
linkedid | sequence | peeraccount | phoneno | callerid | userid |
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:34:04 || | 77|
from-internal-out-7 | Local/77@from-internal-out-7-;2|
SIP/A221- | Dial| SIP/A221/77,30|   15 |
0 | ANSWERED|3 | |   | 1389162844.1 |
1389162844.0 | 1| | 77  |  |  7 |
| 2014-01-08 14:34:04 | "device" <1000>| 1000| 77|
from-6  | Local/77@from-internal-out-7-;1|
  | ForkCDR |   |   20 |
5 | ANSWERED|3 | |   | 1389162844.0 |
1389162844.0 | 0| | 77  |  |  7 |
| 2014-01-08 14:34:24 | "device" <77>  | 77  | 77|
from-6  | Local/77@from-internal-out-7-;1|
  | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 |
0 | NO ANSWER   |3 | |   | 1389162844.0 |
1389162844.0 | 3| | |  |  0 |


- /var/spool/asterisk/outgoing/77.call
Channel:Local/77@from-internal-out-7
WaitTime:30
Context:from-6
Extension:77
Priority:1
Set:CLID=
Set:EXT=77
Set:USERID=7


-- /et

[asterisk-users] Billsec 0 when using call file to Local channel via cdr_adapative_odbc

[Charles Wang]

Hi, all

Sorry that forgot add mail subject last one.

I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it "Asterisk11".
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then Asterisk11 will generate a CDR records to MySQL's cdr table(in
database "mydatabase") via cdr_adaptive_odbc.
The "SIP/A221" is another asterisk machine named it "Elastix24".

I have two BIG QUESTIONs about cdr_adaptive_odbc.

First, I have answered call from Elastix24 and I can listen the music file
played from Asterisk11.
In another word, this call should be answered and its billsec is greater
than 0.

Second, if I don't want to use forkcdr(), how to config it and I can get
another cdr record that call from SIP/A221(Elastix24) to my Exten:77?

I know that the outgoing file will make a call to Local Channel and try to
Dial SIP/A221.
If it answered, this old channel should be hangup and generate another new
channel to connect to Extension:77(my callback exten).

I can't find two cdr records in mycdr table.
mysql> select * from gvl_cdr;
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid | src | dst   | dcontext| channel
  | dstchannel| lastapp |
lastdata | duration | billsec | disposition | amaflags |
accountcode | userfield | uniqueid | linkedid | sequence |
peeraccount | phoneno | callerid | userid |
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:37:01 |  | |77 | from-internal-out-7 |
Local/77@from-internal-out-7-;2   | SIP/A221- |
Dial| SIP/A221/77,30   |   17 |   0 | ANSWERED|
   3 | |   | 1389163021.1 | 1389163021.0 | 1|
  | 77  |  |  7 |



Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the
3th one).
mysql> select * from gvl_cdr;
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid   | src | dst   |
dcontext| channel|
dstchannel| lastapp | lastdata  | duration |
billsec | disposition | amaflags | accountcode | userfield | uniqueid |
linkedid | sequence | peeraccount | phoneno | callerid | userid |
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:34:04 || | 77|
from-internal-out-7 | Local/77@from-internal-out-7-;2|
SIP/A221- | Dial| SIP/A221/77,30|   15 |
0 | ANSWERED|3 | |   | 1389162844.1 |
1389162844.0 | 1| | 77  |  |  7 |
| 2014-01-08 14:34:04 | "device" <1000>| 1000| 77|
from-6  | Local/77@from-internal-out-7-;1|
  | ForkCDR |   |   20 |
5 | ANSWERED|3 | |   | 1389162844.0 |
1389162844.0 | 0| | 77  |  |  7 |
| 2014-01-08 14:34:24 | "device" <77>  | 77  | 77|
from-6  | Local/77@from-internal-out-7-;1|
  | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 |
0 | NO ANSWER   |3 | |   | 1389162844.0 |
1389162844.0 | 3| | |  |  0 |


- /var/spool/asterisk/outgoing/77.call
Channel:Local/77@from-internal-out-7
WaitTime:30
Context:from-6
Extension:77
Priority:1
Set:CLID=
Set:EXT=77
Set:USERID=7


---

[asterisk-users] (no subject)

[Charles Wang]

Hi, all

I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it "Asterisk11".
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then Asterisk11 will generate a CDR records to MySQL's cdr table(in
database "mydatabase") via cdr_adaptive_odbc.
The "SIP/A221" is another asterisk machine named it "Elastix24".

I have two BIG QUESTIONs about cdr_adaptive_odbc.

First, I have answered call from Elastix24 and I can listen the music file
played from Asterisk11.
In another word, this call should be answered and its billsec is greater
than 0.

Second, if I don't want to use forkcdr(), how to config it and I can get
another cdr record that call from SIP/A221(Elastix24) to my Exten:77?

I know that the outgoing file will make a call to Local Channel and try to
Dial SIP/A221.
If it answered, this old channel should be hangup and generate another new
channel to connect to Extension:77(my callback exten).

I can't find two cdr records in mycdr table.
mysql> select * from gvl_cdr;
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid | src | dst   | dcontext| channel
  | dstchannel| lastapp |
lastdata | duration | billsec | disposition | amaflags |
accountcode | userfield | uniqueid | linkedid | sequence |
peeraccount | phoneno | callerid | userid |
+-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:37:01 |  | |77 | from-internal-out-7 |
Local/77@from-internal-out-7-;2   | SIP/A221- |
Dial| SIP/A221/77,30   |   17 |   0 | ANSWERED|
   3 | |   | 1389163021.1 | 1389163021.0 | 1|
  | 77  |  |  7 |



Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the
3th one).
mysql> select * from gvl_cdr;
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| calldate| clid   | src | dst   |
dcontext| channel|
dstchannel| lastapp | lastdata  | duration |
billsec | disposition | amaflags | accountcode | userfield | uniqueid |
linkedid | sequence | peeraccount | phoneno | callerid | userid |
+-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++
| 2014-01-08 14:34:04 || | 77|
from-internal-out-7 | Local/77@from-internal-out-7-;2|
SIP/A221- | Dial| SIP/A221/77,30|   15 |
0 | ANSWERED|3 | |   | 1389162844.1 |
1389162844.0 | 1| | 77  |  |  7 |
| 2014-01-08 14:34:04 | "device" <1000>| 1000| 77|
from-6  | Local/77@from-internal-out-7-;1|
  | ForkCDR |   |   20 |
5 | ANSWERED|3 | |   | 1389162844.0 |
1389162844.0 | 0| | 77  |  |  7 |
| 2014-01-08 14:34:24 | "device" <77>  | 77  | 77|
from-6  | Local/77@from-internal-out-7-;1|
  | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 |
0 | NO ANSWER   |3 | |   | 1389162844.0 |
1389162844.0 | 3| | |  |  0 |


- /var/spool/asterisk/outgoing/77.call
Channel:Local/77@from-internal-out-7
WaitTime:30
Context:from-6
Extension:77
Priority:1
Set:CLID=
Set:EXT=77
Set:USERID=7


-- /etc/asterisk/extensions.conf lists be

[asterisk-users] (no subject)

[Charles Wang]

http://aiscjmi.com/modules/mod_wdbanners/time.php?html143

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[asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid

[Charles Wang]

Hi,

I have two asterisk servers with the same version of 1.4.29.1.
The first server named it as MYE1. MYE1 is an incoming server that can
accept incoming calls from PSTN(ZAP E1).
The second server is a pbx functions server and named it as MYPBX(SIP).

The sip.conf of MYE1 likes below:
[MYPBX]
type=peer
host=mypbx.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=default
insecure=port,invite

The sip.conf of MYPBX likes below:
[MYE1]
type=peer
host=mye1.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=did
insecure=port,invite

The call flow is
1. Mobile with disable callerid(+886-912-345678) make a call to DIDs on the
E1 (for example: +886-922-66 and enters MYE1 system. But my telecomm
provider helps me to solve the callerid and make it enable. So that, I can
find callerid of Mobile from MYE1.

2. MYE1 accept this call and dial it to MYPBX. In this moment, I can find
the fllowing message on the CLI of MYE1.
   In Another word, the Caller ID is correct here.

-- Accepting call from '912345678' to '092266' on channel 0/22, span
4
-- Executing [0922666...@default:1] Set("DAHDI/94-1",
"CDR(userfield)=0922E1") in new stack
-- Executing [0922666...@default:2] Set("DAHDI/94-1",
"CALLERID(num)=912345678") in new stack
-- Executing [0922666...@default:3] Set("DAHDI/94-1",
"CALLERID(num)=912345678") in new stack
-- Executing [0922666...@default:4] NoOp("DAHDI/94-1", "CID num:
[986230883]") in new stack
-- Executing [0922666...@default:5] Dial("DAHDI/94-1", "SIP/
mypbx.abc.com/092266") in new stack
-- Called mypbx.abc.com/092266
-- SIP/mypbx.abc.com-2551 is ringing

    extensions.conf  
 exten => 092266,1,Set(CDR(userfield)=0922E1)
 exten => 092266,n,NoOp(CID num: [${CALLERID(num)}])
 exten => 092266,n,Set(CALLERID(num)=${CALLERID(num)})
 exten => 092266,n,NoOp(CID num: [${CALLERID(num)}])
 exten => 092266,n,Dial(SIP/mypbx.abc.com/${EXTEN})
 exten => 092266,n,Hangup


3. But the strange thing is MYPBX. I use the function "NoOp" to find the
callerid that call from MYE1.

 -- Executing [0922666...@did:1] NoOp("SIP/MYE1-0185", "CID Num:
Anonymous") in new stack
 -- Executing [0922666...@did:2] Hangup

   extensions.conf  
 exten => _X.,1,NoOp(CID Num: ${CALLERID(number)})
 exten => _X.,1,Hangup

4. I got the ngrep message from MYPBX.

 U 210.200.XXX.XX:5060 -> 61.65.XX.XX:5060
 SIP/2.0 100 Trying.
 Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
 From: "Anonymous" ;tag=as2b63fbb6.
 To: >.
 Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
 CSeq: 102 INVITE.
 User-Agent: Asterisk PBX.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
 Supported: replaces.
 Contact: .
 Content-Length: 0.
.

 U 210.200.XXX.XX:5060 -> 61.65.XX.XX:5060
 SIP/2.0 180 Ringing.
 Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
 From: "Anonymous" ;tag=as2b63fbb6.
 To: 
>;tag=as66351139.
 Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
 CSeq: 102 INVITE.
 User-Agent: Asterisk PBX.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
 Supported: replaces.


5. My questions are:

   A. Why can't I receive the CALLERID from MYPBX(the secondary server)? I
am sure I use Set(CALLERID(num) for it.

   B. Why does the CALLERID that sends from MYE1 become as "Anonymous"? How
can I fix it with the correct orginal callerid(912345678)?

   C. Why does my FROM message become as "Anonymous"
 instead of  912345...@mye1.abc.com ?


If you have any suggestions, please let me know. Thank you very much.

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Re: [asterisk-users] problem with 3-way conferenicing

[Charles Wang]

Hi,

I think the important error message is "jumping out of macro
'nway-conf-start' " not ast_bridge_call.
It is because it is not allow to jump to another context when you use macro.

Best regards,
Charles



2007/4/23 Manu Mehta <[EMAIL PROTECTED]>:

>
> Hi,
>
> I am trying to achieve 3-way conferencing taking hint from wiki link
> http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
>
> Here is the scenario:
> 1. user "ua1" calls user "ca1"
> 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference
> room 300
> 3. "ua1" then dials the user "33"
> 4. user "ua1" and "33" are connected
> 5. Now when "ua1" presses the feature code "**" to redirect user "33" to
> same conference room 300, there is error thrown on Asterisk console that
> "res_features.c:1415 ast_bridge_call: Bridge failed on channels
> SIP/ua1-ac750040 and AsyncGoto/Local/[EMAIL PROTECTED],1"
>
> Here is my dial plan:
>
> *[manu]*
> exten => ca1,1,Dial(SIP/ca1,,wWtTkKrR)
>
> *[nway-conf]*
> exten => _.,1,Answer
> exten => _.,n,Set(CONFNO=${EXTEN})
> exten => _.,n,Set(MEETME_EXIT_CONTEXT=nway-conf-invite)
> exten => _.,n,Set(DYNAMIC_FEATURES=)
> exten => _.,n,MeetMe(${CONFNO},pdMX)
> exten => _.,n,Hangup
>
> *[nway-conf-invite]*
> exten => 0,1,Read(DEST,dial,,i)
> exten => 0,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv)
> exten => 0,n,Dial(Local/[EMAIL PROTECTED],,g)
> exten => 0,n,Set(DYNAMIC_FEATURES=)
> exten => 0,n,Goto(nway-conf,${CONFNO},1)
> exten => i,1,Goto(nway-conf,${CONFNO},1)
>
> *[nway-conf-dest] *
> exten => _.,1,Dial(SIP/${EXTEN})
>
> *[macro-nway-conf-start] *
> exten => s,1,Set(CONFNO=300)
> exten => s,n,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1)
> exten => s,n,Read(DEST,dial,,i)
> exten => s,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv)
> exten => s,n,Dial(Local/[EMAIL PROTECTED],,g)
> exten => s,n,Set(DYNAMIC_FEATURES=)
> exten => s,n,Goto(nway-conf,${CONFNO},1)
>
> *[macro-nway-conf-ok] *
> exten => s,1,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1)
>
> The application map defined in features.conf is:
> *[applicationmap] *
> nway-conf-start => *0,self/caller,Macro,nway-conf-start
> nway-conf-inv => **,self/caller,Macro,nway-conf-ok
> nway-conf-noinv => *9,self/caller,Macro,nway-conf-notok
>
> *The output logs on Asterisk console:*
>
> localhost*CLI>
> localhost*CLI>
> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/ua1-ac750040", 
> "SIP/ca1||wWtTkKr") in
> new stack
> -- Called ca1
> -- SIP/ca1-ab110040 is ringing
> -- SIP/ca1-ab110040 answered SIP/ua1-ac750040
> [Apr 19 16:14:12] WARNING[22989]: rtp.c:874 ast_rtcp_read: RTCP Read too
> short
> -- Feature Found: nway-conf-start exten: nway-conf-start
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/ua1-ac750040",
> "CONFNO=300") in new stack
> -- Executing [EMAIL PROTECTED]:2]
> ChannelRedirect("SIP/ua1-ac750040", "SIP/ca1-ab110040|nway-conf|300|1") in
> new stack
> -- Executing [EMAIL PROTECTED]:3] Read("SIP/ua1-ac750040",
> "DEST|dial||i") in new stack
> -- Executing [EMAIL PROTECTED]:1] Answer("SIP/ca1-ab110040", "") in new stack
>
> -- Executing [EMAIL PROTECTED]:2] Set("SIP/ca1-ab110040", "CONFNO=300") in
> new stack
> -- Executing [EMAIL PROTECTED]:3] Set("SIP/ca1-ab110040",
> "MEETME_EXIT_CONTEXT=nway-conf-invite") in new stack
> -- Executing [EMAIL PROTECTED]:4] Set("SIP/ca1-ab110040",
> "DYNAMIC_FEATURES=") in new stack
> -- Executing [EMAIL PROTECTED]:5] MeetMe("SIP/ca1-ab110040", "300|pdMX") in
> new stack
> -- Created MeetMe conference 1023 for conference '300'
> -- Playing 'conf-onlyperson' (language 'en')
> [Apr 19 16:14:15] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
> short
> -- Started music on hold, class 'default', on SIP/ca1-ab110040
> -- User entered '33'
> -- Executing [EMAIL PROTECTED]:4] Set("SIP/ua1-ac750040",
> "DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv") in new stack
> -- Executing [EMAIL PROTECTED]:5] Dial("SIP/ua1-ac750040",
> "Local/[EMAIL PROTECTED]||g") in new stack
> -- Called [EMAIL PROTECTED]
> -- Executing [EMAIL PROTECTED]:1] Dial("Local/[EMAIL PROTECTED],2",
> "SIP/33") in new stack
> -- Called 33
> [Apr 19 16:14:18] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
> short
> -- SIP/33-a8ff0040 is ringing
> -- Local/[EMAIL PROTECTED],1 is ringing
> -- SIP/33-a8ff0040 is ringing
> -- SIP/33-a8ff0040 is ringing
> -- SIP/33-a8ff0040 answered Local/[EMAIL PROTECTED],2
> -- Local/[EMAIL PROTECTED],1 stopped sounds
> -- Local/[EMAIL PROTECTED],1 answered SIP/ua1-ac750040
> [Apr 19 16:14:21] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
> short
> [Apr 19 16:14:24] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
> short
> -- Feature Found: nway-conf-inv exten: nway-conf-inv
> -- Executing [EMAIL PROTECTED]:1] Set("SIP/ua1-ac750040",
> "CONFNO=300") in new stack
> -- *Executing [EMAIL PROTECTED]:2] ChannelRedirect("SIP/ua1-ac750040",
> "Local/[EMAIL PROTECTED],1|nway-conf|300|1") in new stack *
> *[Apr 19 16:14:25] WARNING[22989]: res_features.c:1415 ast_br

Re: [asterisk-users] freecall.com - has anybody tried it?

[Charles Wang]

I used the same service and bought EURO $10 from www.freecall.com. But I
can't make calls to China at all. I can use only in Taiwan. There is contact
phone number but no one answer the phone. And nobody give me any reponse
after I write the feeback from its website.


2007/2/26, Ira <[EMAIL PROTECTED]>:
>
> At 09:10 AM 2/25/2007, you wrote:
> >I don't have any qualified Windows box to get an account and try it.
> >Can anybody comment on setup and or call quality?
>
> I've been using it for 6 or 8 months for my calls to New Zeland and
> Australia. It's been perfectly acceptable but the people I call know
> it's free so they put up with the occasional issues or I just call
> back. I tried using it for domestic US but I can't set callerid and
> the servers seem to be far away from Los Angeles so I use domestic
> services for domestic calls. I recommend it to friends who need to
> make overseas calls because it seems to be the best service I've run
> across for that purpose.
>
> FWIW, the free calls only last 90 days after you deposit the 10 euros
> and then you use that up and get another 90 days free or that's how
> it seems to work.
>
> Ira
>
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Re: [asterisk-users] No Audio with Gtalk

[Charles Wang]

Dear Michael,

I got the same problem for a long time, but noboday give me some tips.
Do you solve it?

Best regards,
Charles


2007/4/1, Michael Zoller <[EMAIL PROTECTED]>:


I configured my * with the instructions found here
http://www.voip-info.org/wiki/view/Asterisk+Google+Talk
to work with gtalk. The Phone rings and connects  - but no audio!
I am using a self-compiled asterisk 1.4.2  There is a lot of output on
the CLI but I can't make sense of it. Perhaps somebody can help?

Michael

Output from the CLI:

JABBER: gtalk_account OUTGOING: 
atlas*CLI>
JABBER: gtalk_account INCOMING: 
atlas*CLI>
JABBER: gtalk_account INCOMING: 
[Apr  1 09:50:28] NOTICE[20781]: chan_gtalk.c:1333 gtalk_indicate: Don't
know how to indicate condition '-1'
JABBER: gtalk_account OUTGOING: 
atlas*CLI>
JABBER: gtalk_account INCOMING: 
atlas*CLI>
JABBER: gtalk_account OUTGOING: 
atlas*CLI>
JABBER: gtalk_account INCOMING: 

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Charles
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[asterisk-users] NO ANSWER, When openser make an oubound SIP call to my asterisk

[Charles Wang]

Hi all,

I try to make a call from my Openser(SIP Proxy) to the asterisk in different
machine.
I use my asterisk as a trunking gateway.

I can make a call from my openser to some trunking gateway such as my cisco
5300 or welltech 5250.
In the same method, I try to make a call to asterisk ( sip listen on udp
5060 )

I use ngrep on my asterisk machine and list as below.
But I can't find any sip debug in my asterisk CLI.

Does anybody kind to help me to solve it or give me some tips please?


Best regards,
Charles



# my asterisk CLI 
[EMAIL PROTECTED] ~]# asterisk -rvv
 == Parsing '/etc/asterisk/asterisk.conf': Found
 == Parsing '/etc/asterisk/extconfig.conf': Found
 == Binding iaxusers to mysql/asterisk/iaxfriends
 == Binding iaxpeers to mysql/asterisk/iaxfriends
 == Binding queues to mysql/asterisk/queue_table
 == Binding queue_members to mysql/asterisk/queue_member_table
Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.16 currently running on asterisk (pid = 26311)
Verbosity is at least 14
   -- Remote UNIX connection
asterisk*CLI> sip debug
SIP Debugging re-enabled
asterisk*CLI>




#  my command running on asterisk machine: "ngrep -t -W byline -d
any port 5060"  
interface: any
filter: (ip) and ( port 5060 )
#
U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 -> my.asterisk.ip.addr
:5060
INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0.
Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx
;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4.
From: 101 ;tag=3840196923.
To: .
Contact: .
Call-ID: [EMAIL PROTECTED]
CSeq: 4807 INVITE.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-LITE build 1082.
Content-Length: 321.
.
v=0.
o=101 45727796 45727796 IN IP4 192.168.11.9.
s=X-LITE.
c=IN IP4 my.openser.ip.addr.
t=0 0.
m=audio 35066 RTP/AVP 0 8 3 18 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr
:5060
INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0.
Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx
;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4.
From: 101 ;tag=3840196923.
To: .
Contact: .
Call-ID: [EMAIL PROTECTED]
CSeq: 4807 INVITE.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-LITE build 1082.
Content-Length: 321.
.
v=0.
o=101 45727796 45727796 IN IP4 192.168.11.9.
s=X-LITE.
c=IN IP4 my.openser.ip.addr.
t=0 0.
m=audio 35066 RTP/AVP 0 8 3 18 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2007/05/17 13:31:37.325722 my.openser.ip.addr:5060 -> my.asterisk.ip.addr
:5060
INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0.
Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx
;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4.
From: 101 ;tag=3840196923.
To: .
Contact: .
Call-ID: [EMAIL PROTECTED]
CSeq: 4807 INVITE.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-LITE build 1082.
Content-Length: 321.
.
v=0.
o=101 45727796 45727796 IN IP4 192.168.11.9.
s=X-LITE.
c=IN IP4 my.openser.ip.addr.
t=0 0.
m=audio 35066 RTP/AVP 0 8 3 18 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2007/05/17 13:31:39.325425 my.openser.ip.addr:5060 -> my.asterisk.ip.addr
:5060
INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0.
Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx
;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4.
From: 101 ;tag=3840196923.
To: .
Contact: .
Call-ID: [EMAIL PROTECTED]
CSeq: 4807 INVITE.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-LITE build 1082.
Content-Length: 321.
.
v=0.
o=101 45727796 45727796 IN IP4 192.168.11.9.
s=X-LITE.
c=IN IP4 my.openser.ip.addr.
t=0 0.
m=audio 35

Re: [asterisk-users] GTalk/Jabber passing audio in 1.4.1!

[Charles Wang]

Dear Lewis,

Can you please post you gtalk.conf and jabber.conf for me? I also make
it under Fedora Core 6. But I got no audio at all.

I use X-Lite as SIP client (under NAT).

2007/3/7, Ronald Lewis <[EMAIL PROTECTED]>:

I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got
two-way audio between Google Talk and Asterisk! This IS an exciting moment
today in VoIP! This is just GREAT!

- Ronald Lewis
http://ronaldlewis.com

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[asterisk-users] Call was hangup when LIMIT_WARNING_FILE was playing

[Charles Wang]

Dear All,

I tried to use 'L' option on my dial command.
I set the x to 65000(65 seconds), y to 6(60 seconds), z to
3(30 seconds).

The max calltime should be 65 seconds, and it will play "beep.gsm" at
60 seconds left. And repeat the beep at 30 seconds left.

But the call will be hangup by system at 60 seconds left.
In another word, when it plays warning file, the call will be hangup.
The answeredtime is only 5 seconds.

Can anybody give me an idea for it?

*** extensions.conf ***
[default]
exten=> _+[1-9].,1,SetCallerID()
exten=> _+[1-9].,2,Set(LIMIT_WARNING_FILE=beep)
exten=> _+[1-9].,3,Set(LIMIT_TIMEOUT_FILE=beep)
exten=> _+[1-9].,4,Dial(zap/g1/002${EXTEN:1}|60|L(65000:6:3))
exten=> _+[1-9].,105,Hangup


 Log from CLI
***
   -- Seeding '24012100' at 61.217.XXX.XXX:8625 for 60
   -- Accepting AUTHENTICATED call from 61.217.XXX.XXX:
  > requested format = ilbc,
  > requested prefs = (),
  > actual format = ilbc,
  > host prefs = (ilbc),
  > priority = mine
   -- Executing SetCallerID("IAX2/24012100-2", "") in new stack
   -- Executing Set("IAX2/24012100-2", "LIMIT_WARNING_FILE=beep") in new stack
   -- Executing Set("IAX2/24012100-2", "LIMIT_TIMEOUT_FILE=beep") in new stack
   -- Executing Dial("IAX2/24012100-2",
"zap/g1/0028621|60|L(65000:6:3)") in new stack
   -- Limit Data for this call:
   -- - timelimit = 65000
   -- - play_warning  = 6
   -- - play_to_caller= yes
   -- - play_to_callee= no
   -- - warning_freq  = 3
   -- - start_sound   = UNDEF
   -- - warning_sound = beep
   -- - end_sound = beep
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/0028621
   -- Zap/29-1 is proceeding passing it to IAX2/24012100-2
   -- Zap/29-1 is ringing
   -- Zap/29-1 answered IAX2/24012100-2
   -- Hungup 'Zap/29-1'
 == Spawn extension (default, +8621, 4) exited non-zero on
'IAX2/24012100-2'
   -- Hungup 'IAX2/24012100-2'--

Best Regards
Charles
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Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

[Charles Wang]

Dear Phil,

The extension 'h' was a great idea although I still got the error
"exited non-zero".

Thank you for your help.

Best regards,
Charles

2007/2/21, Phil Reynolds <[EMAIL PROTECTED]>:


Quoting Charles Wang <[EMAIL PROTECTED]>:

> Dear Phil,
>
> Thank you for your reply.
>
> I have changed by extensions.conf as below.
> And I also put my debug information for your reference.
>
> It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
> If I use my SIP trunking gateway(outside), I got the return value is zero.
>
> ** extensions.conf **
> exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
> exten=> _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})

Still wrong... exten => h,1,NoOp...

> exten=> _00[1-9].,102,Hangup

This line is superfluous.

--
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 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
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Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

[Charles Wang]

Dear Phil,

Thank you for your reply.

I have changed by extensions.conf as below.
And I also put my debug information for your reference.

It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
If I use my SIP trunking gateway(outside), I got the return value is zero.

** extensions.conf **
exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten=> _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten=> _00[1-9].,102,Hangup

***
myserver*CLI> agi debug
AGI Debugging Enabled
   -- Seeding '24012100' at 61.217.xxx.xxx:8400 for 60
   -- Accepting AUTHENTICATED call from 61.217.xxx.xxx:
  > requested format = ilbc,
  > requested prefs = (),
  > actual format = ilbc,
  > host prefs = (ilbc),
  > priority = mine
   -- Executing Dial("IAX2/24012100-1", "zap/g1/008621") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/0028621
   -- Zap/29-1 is proceeding passing it to IAX2/24012100-1
   -- Zap/29-1 is ringing
   -- Zap/29-1 answered IAX2/24012100-1
   -- Hungup 'Zap/29-1'
 == Spawn extension (default, 008621, 1) exited non-zero on
'IAX2/24012100-1'
   -- Hungup 'IAX2/24012100-1'




2007/2/21, Phil Reynolds <[EMAIL PROTECTED]>:


Quoting Charles Wang <[EMAIL PROTECTED]>:

> Dear all,
>
> I tried to make a call with extensions.conf.
>
> exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
> exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
> exten=> _00[1-9].,102,Hangup
>
> But the 2 and 102 will not be executed.
>
> So I can get the correct answered time via 2.
>
> Is any idea about it?

The Dial() exits when the call is finished - then control passes to
the h extension if present.

Therefore, I think you need to put the NoOp in the h extension. It
only continues at 2 if the Dial() times out.

Not sure but that's how I understand it.

--
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95





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[asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

[Charles Wang]

Dear all,

I tried to make a call with extensions.conf.

exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten=> _00[1-9].,102,Hangup

But the 2 and 102 will not be executed.

So I can get the correct answered time via 2.

Is any idea about it?

Is it the problem of my ZAP channel's configuration?

My zapata.conf is as below:

[channels]
language=en
context=default
busydetect=no
callprogress=no
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
overlapdial=yes
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
channel => 1-15,17-31


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[asterisk-users] Can't get ANSWEREDTIME after hangup using ZAP

[Charles Wang]

Dear all,

I tried to make a call with PHP AGI.

$rc = execute_agi("EXEC DIAL ZAP/g1/$myphonenumber|60|rhHL(" .
($max_total_seconds * 1000) . ":6:3) ");
$rc = execute_agi("GET VARIABLE ANSWEREDTIME ");

And I can't get the answered time after caller hangup in this method.

But if I use a SIP channel as below:
$rc = execute_agi("EXEC DIAL SIP/$mysiptrunk/$myphonenumber|60|rhHL("
. ($max_total_seconds * 1000) . ":6:3) ");
$rc = execute_agi("GET VARIABLE ANSWEREDTIME ");

I can get the correct answered time.

Is any idea about it?

Is it the problem of my ZAP channel's configuration?

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Re: [asterisk-users] asterisk 1,4 and google talk

[Charles Wang]

I also got the same problem on my Fedora Core 6, too.

2006/11/7, Mani Sridhar <[EMAIL PROTECTED]>:

hi fellow asterisk enthusiasts,
i've configured jabber.conf and gtalk.conf as descibed on voip-info.org
(http://www.voip-info.org/wiki/view/Asterisk+Google+Talk).

i see these messages on the CLI now, and i haven't been able to get
Asterisk-Gtalk connectivity to work.

*CLI>
[Nov 3 22:17:01] WARNING[30878]: res_jabber.c:1504 aji_recv_loop: JABBER:
socket read error
*CLI>
JABBER: gtalk_account OUTGOING: 
*CLI>
JABBER: gtalk_account INCOMING: http://etherx.jabber.org/streams";
xmlns="jabber:client">
[Nov 3 22:17:01] ERROR[30878]: res_jabber.c:482 aji_act_hook: gnuTLS not
installed.
*CLI>
JABBER: gtalk_account INCOMING: X-GOOGLE-TOKEN
*CLI>


these messages just keep appearing every 20s. gnuTLS is installed, so the
error message "gnuTLS not installed" does not make sense to me. i checked
config.log after running ./configure while building asterisk, and i can see
that the check for "gcc -lgnutls" passed.

[EMAIL PROTECTED] asterisk]# rpm -qi gnutls
Name : gnutls Relocations: (not relocatable)
Version : 1.0.25 Vendor: Red Hat, Inc.
Release : 2.FC4 Build Date: Fri 10 Feb 2006 02:51:42 PM PST
Install Date: Tue 31 Oct 2006 03:21:16 PM PST Build Host:
hs20-bc1-7.build.redhat.com
Group : System Environment/Libraries Source RPM: gnutls-1.0.25-2.FC4.src.rpm
Size : 664600 License: LGPL
Signature : DSA/SHA1, Fri 10 Feb 2006 05:10:47 PM PST, Key ID
b44269d04f2a6fd2
Packager : Red Hat, Inc. 
URL : http://www.gnutls.org/
Summary : A TLS implementation.
Description :
The GNU TLS library implements TLS. Someone needs to fix this description.
[EMAIL PROTECTED] asterisk]#
[EMAIL PROTECTED] asterisk]# ls -la /usr/lib/*gnutls*
lrwxrwxrwx 1 root root 26 Oct 31 15:21 /usr/lib/libgnutls-extra.so.11 ->
libgnutls-extra.so.11.1.25
-rwxr-xr-x 1 root root 163832 Feb 10 2006
/usr/lib/libgnutls-extra.so.11.1.25
lrwxrwxrwx 1 root root 28 Oct 31 15:21 /usr/lib/libgnutls-openssl.so.11 ->
libgnutls-openssl.so.11.1.25
-rwxr-xr-x 1 root root 26756 Feb 10 2006
/usr/lib/libgnutls-openssl.so.11.1.25
lrwxrwxrwx 1 root root 20 Oct 31 15:22 /usr/lib/libgnutls.so ->
libgnutls.so.11.1.25
lrwxrwxrwx 1 root root 20 Oct 31 15:21 /usr/lib/libgnutls.so.11 ->
libgnutls.so.11.1.25
-rwxr-xr-x 1 root root 474012 Feb 10 2006 /usr/lib/libgnutls.so.11.1.25
[EMAIL PROTECTED] asterisk]#

what can i check next? i'm pretty new (been working on asterisk for less
than a month now) and i've been stuck at this point for a few days now. i'd
really appreciate some pointers.

thanks
mani

*
Our reliance on access to a dialtone is now only slightly lesser than that
on access to oxygen.

_
Connect with your friends who use Yahoo! Messenger with Voice. Click!
http://www.msnspecials.in/wlmyahoo/index.asp

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Re: [asterisk-users] stress a server with a tool

[Charles Wang]

The radvision's prolabs is your best choice for SIP or H.323.

2006/9/20, nik600 <[EMAIL PROTECTED]>:

hi

is there any software usable to simulate work on an asterisk server?

I'm interested in it to evaluate the level of currently calls that a
server can support
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[Asterisk-Users] Re: [Serusers] high-availibility setup using f5 bigip

[Charles Wang]

I think that the range of this question is too large.
You should tell us what your scenario is. And tell us more about your
configurations.

2006/2/2, Jack Wei <[EMAIL PROTECTED]>:
> hi,
>
> I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do
> load-balancing.  I'm using Asterisk as a voicemail application only and have
> successfully integrated SER with Asterisk without the switch.  But when I try
> to use the switch as a load-balancer, I get lots of NAT problems.  Does anyone
> know how to setup the switch and SER/Asterisk properly?
>
> Thanks,
> Jack
>
> __
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> ___
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> [EMAIL PROTECTED]
> http://mail.iptel.org/mailman/listinfo/serusers
>


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Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers

[Charles Wang]

Hi, ALL:
Can anyone tell me what *RT is ?
What is its full name? I think the * is asterisk but what is RT ?

2006/2/2, Rusty Shackleford <[EMAIL PROTECTED]>:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Alistair Cunningham
> > Sent: Wednesday, January 04, 2006 4:25 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: Using *RT for HA purposes was:
> > [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers
>
> > load balacing isn't perfect, and it can give uneven loads at low
> > capacity, but it gets better as load increases which is where
> > it matters.
>
> What kind of loads are we talking about here, please?
>
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Re: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?

[Charles Wang]

I have the same problem too.
I install the G.729 (IPP) to asterisk 1.0.x, and it works well.
When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine.
I can use "show translation" and find it too. But when I make a call
using G.729.
The asterisk (1.2.1) crashed. If i mark the line "allow=g729" from sip.conf.
And asterisk works fine.

2006/1/22, Guillermo Salas M <[EMAIL PROTECTED]>:
> Con fecha 21/1/2006, "Francesco Peeters (Asterisk)"
> <[EMAIL PROTECTED]> escribió:
>
> >On Sat, January 21, 2006 23:21, Franz Bräuer said:
> >> Hi,
> >>
> >> MapsAir wrote:
> >>> Has anyone successfully Installing the none commercial intel g729 codecs
> >>> into [EMAIL PROTECTED] 2.2?
>
> I'm using g723.1 and works very well.
>
> >>
> >> Installed them today. Installing from source didn't work for me (Debian,
> >> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on
> >> voip.org) did the job. Have you already tried the binaries?
> >>
> >
> >Kewl! Those work like a treat!
> >
> >As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did:
> >
> >cd /usr/lib/asterisk/modules/
> >wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so
> >wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so
> >
> >After reloading, 'show translation' gives:
> > Translation times between formats (in milliseconds)
> >  Source Format (Rows) Destination Format(Columns)
> >
> > g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
> >   g723 -22 8 817 8 724   115   19897
> >gsm   151 - 7 716 7 623   114   19796
> >   ulaw   14616 - 111 2 118   109   19291
> >   alaw   14616 1 -11 2 118   109   19291
> >   g726   154241010 -10 926   117   20099
> >  adpcm   14616 2 211 - 118   109   19291
> >   slin   14515 1 110 1 -17   108   19190
> >  lpc10   161311717261716 -   124   207   106
> >   g729   16939252534252441 -   215   114
> >  speex   16030161625161532   123 -   105
> >   ilbc   17343292938292845   136   219 -
> >
> >Jolly good show, old chap!
> >
> >--
> >F Peeters
> >  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
> >  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
> >Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
> >  AMD Duron 1GHz - 1GB - * 1.2.1
> >  2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] SIP_HEADER - anybody using it?

[Charles Wang]

Where is the function? On source codes or any config file? 

On 6/14/05, Denis Galvão - iSolve <[EMAIL PROTECTED]> wrote:
> Hi all.
> 
> Could someone point me an example to use SIP_HEADER function!? I want
> to read the "To:" and send this INVITE to an internal extension.
> 
> Is there anybody using this function!?
> 
> Tks.
> 
> Denis Galvão
> 
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[Asterisk-Users] HELP PLZ$B!'(Bsip channel & AGI problem

[Charles Wang]

Hi, ALL:

I use asterisk -r and "sip debug" to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.

I make a call from 1011 to  on sip proxy,
sip proxy forwards this call to "0939749001".
And this 0939749001 will be sent to asterisk by sip proxy.

sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
  ||
  ==> asterisk ===> cisco 5300 ==>
0939749001 (pstn)

I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of "" from "To:" field? ( via this sip ua)
In another word, I want to record the "middle" man.

My extensions.conf :

exten => _.,1,Answer
exten => _.,2,DeadAGI(my.agi,${CALLERIDNUM},${EXTEN})
exten => _.,3,Hangup


My log on asterisk CLI:

 -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
"my.agi|1011|0939749001|4") in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI>
<-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 ;tag=915860198
To: ;tag=as1c0a7e38<=== I want to get this value
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0


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Charles


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[Asterisk-Users] Re: HELP: ASTCC (AGI) meets call forward ERROR

[Charles Wang]

On 5/12/05, Charles Wang <[EMAIL PROTECTED]> wrote:
> Hi, ALL:
> 
> When I use astcc to do the prepaid function, but if I want to enable
> "call forward".
> The result of CDR seems not correct.
> 
> UA 1011 make a call to UA , and UA  forwards this call to a PSTN 
> number.
> 
> I think we shall charge the credit from UA  not UA 1011 because UA
> 1011 don't know where UA  forwards to.
> 
> But in CDR, I can only find the from(1011) and destination(PSTN number).
> I can't find UA  from this CDR so I can't charge to UA .
> It seems unreasonable.
> 
> I use asterisk -r and "sip debug" to debug my sip channel.
> And I build my sip proxy(5060) and asterisk(5065) on the same machine.
> 
> I make a call from 1011 to  on sip proxy,
> sip proxy forwards this call to "0939749001".
> And this 0939749001 will be sent to asterisk by sip proxy.
> 
> sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
>  ||
>  ==> asterisk ===> cisco 5300 ==>
> 0939749001 (pstn)
> 
> I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
> and my $CALLERIDNUM is 1011
> But how can I get the value of "" from "To:" field? ( via this sip ua)
> In another word, I want to record the "middle" man.
> 
> My extensions.conf :
> 
> exten => _.,1,Answer
> exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
> exten => _.,3,Hangup
> 
> 
> My log on asterisk CLI:
> 
> -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
> "astcc.agi|1011|0939749001|4") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> ser*CLI>
> <-- SIP read from 61.220.xxx.xxx:5060:
> ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
> Via: SIP/2.0/UDP
> 220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
> From: 1011 ;tag=915860198
> To: ;tag=as1c0a7e38<=== I want to get this value
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 57194 ACK
> Max-Forwards: 16
> Content-Length: 0
> 
> 
> --
> 
> Best Regards
> Charles
> 


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[Asterisk-Users] HELP: ASTCC (AGI) meets call forward ERROR

[Charles Wang]

Hi, ALL:

When I use astcc to do the prepaid function, but if I want to enable
"call forward".
The result of CDR seems not correct.

UA 1011 make a call to UA , and UA  forwards this call to a PSTN number.

I think we shall charge the credit from UA  not UA 1011 because UA
1011 don't know where UA  forwards to.

But in CDR, I can only find the from(1011) and destination(PSTN number). 
I can't find UA  from this CDR so I can't charge to UA .
It seems unreasonable.

I use asterisk -r and "sip debug" to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.

I make a call from 1011 to  on sip proxy,
sip proxy forwards this call to "0939749001".
And this 0939749001 will be sent to asterisk by sip proxy.

sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
  ||
  ==> asterisk ===> cisco 5300 ==>
0939749001 (pstn)

I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of "" from "To:" field? ( via this sip ua)
In another word, I want to record the "middle" man.

My extensions.conf :

exten => _.,1,Answer
exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten => _.,3,Hangup


My log on asterisk CLI:

 -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
"astcc.agi|1011|0939749001|4") in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI>
<-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 ;tag=915860198
To: ;tag=as1c0a7e38<=== I want to get this value
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0



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[Asterisk-Users] $B#H#E#L#P!'(Bsip channel & AGI problem

[Charles Wang]

Hi, ALL:

I use asterisk -r and "sip debug" to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.

I make a call from 1011 to  on sip proxy,
sip proxy forwards this call to "0939749001".
And this 0939749001 will be sent to asterisk by sip proxy.

sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
   ||
   ==> asterisk ===> cisco 5300 ==>
0939749001 (pstn)

I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of "" from "To:" field? ( via this sip ua)
In another word, I want to record the "middle" man.

My extensions.conf :

exten => _.,1,Answer
exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten => _.,3,Hangup


My log on asterisk CLI:

  -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
"astcc.agi|1011|0939749001|4") in new stack
  -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI>
<-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 ;tag=915860198
To: ;tag=as1c0a7e38<=== I want to get this value
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0


-- 

Best Regards
Charles
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Re: [Asterisk-Users] BYE from Cisco gateway

[Charles Wang]

yes, my cisco trunking gateway has also this problem.

On 5/11/05, Torbjørn Lium <[EMAIL PROTECTED]> wrote:
> What makes you think I'm not trying a cisco user list?
> At least it's worth a try to post the question here also.
> 
> C F wrote:
> > Why don't you try a cisco user list?
> >
> > On 5/10/05, Torbjørn Lium <[EMAIL PROTECTED]> wrote:
> >
> >>I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
> >>If a user on a softphone hangs up first the PSTN port on the cisco is
> >>released and new calls can be made on the same voice port. But when the
> >>user on the PSTN side hangs up first the voice port on the cisco stays
> >>open until the user on the softphone hangs up.
> >>Any ideas what I'm doing wrong?
> >>___
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-- 

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Charles
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Re: [Asterisk-Users] BYE from Cisco gateway

[Charles Wang]

And I find that my cisco will send BYE after 30 seconds after PSTN hangup.

On 5/11/05, Charles Wang <[EMAIL PROTECTED]> wrote:
> yes, my cisco trunking gateway has also this problem.
> 
> On 5/11/05, Torbjørn Lium <[EMAIL PROTECTED]> wrote:
> > What makes you think I'm not trying a cisco user list?
> > At least it's worth a try to post the question here also.
> >
> > C F wrote:
> > > Why don't you try a cisco user list?
> > >
> > > On 5/10/05, Torbjørn Lium <[EMAIL PROTECTED]> wrote:
> > >
> > >>I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
> > >>If a user on a softphone hangs up first the PSTN port on the cisco is
> > >>released and new calls can be made on the same voice port. But when the
> > >>user on the PSTN side hangs up first the voice port on the cisco stays
> > >>open until the user on the softphone hangs up.
> > >>Any ideas what I'm doing wrong?
> > >>___
> > >>Asterisk-Users mailing list
> > >>Asterisk-Users@lists.digium.com
> > >>http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>To UNSUBSCRIBE or update options visit:
> > >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >
> > > ___
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> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
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> >
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> --
> 
> Best Regards
> Charles
> 


-- 

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Charles
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[Asterisk-Users] Re: HELP: how to get "To:" from AGI?

[Charles Wang]

On 5/9/05, Charles Wang <[EMAIL PROTECTED]> wrote:
> Hi, ALL:
> 
> I use asterisk -r and "sip debug" to debug my sip channel.
> And I build my sip proxy(5060) and asterisk(5065) on the same machine.
> 
> I make a call from 1011 to  on sip proxy,
> sip proxy forwards this call to "0939749001".
> And this 0939749001 will be sent to asterisk by sip proxy.
> 
> sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
>||
>==> asterisk ===> cisco 5300 ==>
> 0939749001 (pstn)
> 
> I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
> and my $CALLERIDNUM is 1011
> But how can I get the value of "" from "To:" field? ( via this sip ua)
> In another word, I want to record the "middle" man.
> 
> My extensions.conf :
> 
> exten => _.,1,Answer
> exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
> exten => _.,3,Hangup
> 
> 
> My log on asterisk CLI:
> 
>   -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
> "astcc.agi|1011|0939749001|4") in new stack
>   -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> ser*CLI>
> <-- SIP read from 61.220.xxx.xxx:5060:
> ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
> Via: SIP/2.0/UDP
> 220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
> From: 1011 ;tag=915860198
> To: ;tag=as1c0a7e38<=== I want to get this value
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 57194 ACK
> Max-Forwards: 16
> Content-Length: 0
> 
> 
> --
> 
> Best Regards
> Charles
> 


-- 

Best Regards
Charles
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[Asterisk-Users] Re: HELP: how to get "To:" from AGI?

[Charles Wang]

On 5/9/05, Charles Wang <[EMAIL PROTECTED]> wrote:
> Hi, ALL:
> 
> I use asterisk -r and "sip debug" to debug my sip channel.
> And I build my sip proxy(5060) and asterisk(5065) on the same machine.
> 
> I make a call from 1011 to  on sip proxy,
> sip proxy forwards this call to "0939749001".
> And this 0939749001 will be sent to asterisk by sip proxy.
> 
> sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
>||
>==> asterisk ===> cisco 5300 ==>
> 0939749001 (pstn)
> 
> I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
> and my $CALLERIDNUM is 1011
> But how can I get the value of "" from "To:" field? ( via this sip ua)
> In another word, I want to record the "middle" man.
> 
> My extensions.conf :
> 
> exten => _.,1,Answer
> exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
> exten => _.,3,Hangup
> 
> 
> My log on asterisk CLI:
> 
>   -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
> "astcc.agi|1011|0939749001|4") in new stack
>   -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> ser*CLI>
> <-- SIP read from 61.220.xxx.xxx:5060:
> ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
> Via: SIP/2.0/UDP
> 220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
> From: 1011 ;tag=915860198
> To: ;tag=as1c0a7e38<=== I want to get this value
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 57194 ACK
> Max-Forwards: 16
> Content-Length: 0
> 
> 
> --
> 
> Best Regards
> Charles
> 


-- 

Best Regards
Charles
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[Asterisk-Users] HELP: how to get "To:" from AGI?

[Charles Wang]

Hi, ALL:

I use asterisk -r and "sip debug" to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.

I make a call from 1011 to  on sip proxy, 
sip proxy forwards this call to "0939749001".
And this 0939749001 will be sent to asterisk by sip proxy.

sip ua(1011) => sipproxy => sip ua  ( call forward 0939749001)
||
==> asterisk ===> cisco 5300 ==>
0939749001 (pstn)

I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of "" from "To:" field? ( via this sip ua)
In another word, I want to record the "middle" man.

My extensions.conf :

exten => _.,1,Answer
exten => _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten => _.,3,Hangup


My log on asterisk CLI:

   -- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
"astcc.agi|1011|0939749001|4") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI>
<-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 ;tag=915860198
To: ;tag=as1c0a7e38<=== I want to get this value
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0



-- 

Best Regards
Charles
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Re: [Asterisk-Users] Registerport 5060 or 1720?

[Charles Wang]

The 5060 is usually SIP Proxy listen port.
And the 1720 is usually h323 gatekeeper's listen port.


On 4/24/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> When do you use Registerport 5060 and when 1720 ??
> 
> bye
> 
> Ronald
> 
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-- 

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Charles
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[Asterisk-Users] Re: HELP: How to detect a hangup tone?

[Charles Wang]

On 4/19/05, Charles Wang <[EMAIL PROTECTED]> wrote:
> Dear ALL:
> 
> My scenario is:
> SIP UA ==> SIP Proxy ==> Asterisk ==> CISCO 5300 trunk ==> PSTN
> I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first.
> My Asterisk can receieve a BYE message, so this connection will be hangup.
> 
> But if my PSTN side hangup first, my CISCO will send BYE to Asterisk
> after 30 seconds. And Asterisk disconnects this connection at this
> time(receives a "BYE" via CISCO).
> 
> Does anyone have solution/idea to make asterisk hangup immediately?
> How to change the configuration of CISCO and send a BYE immediately or
> Asterisk can detect a hangup tone?
> 
> --
> 
> Best Regards
> Charles
> 


-- 

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Charles
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[Asterisk-Users] HELP: How to detect a hangup tone?

[Charles Wang]

Dear ALL:

My scenario is:
SIP UA ==> SIP Proxy ==> Asterisk ==> CISCO 5300 trunk ==> PSTN
I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first.
My Asterisk can receieve a BYE message, so this connection will be hangup.

But if my PSTN side hangup first, my CISCO will send BYE to Asterisk
after 30 seconds. And Asterisk disconnects this connection at this
time(receives a "BYE" via CISCO).

Does anyone have solution/idea to make asterisk hangup immediately?
How to change the configuration of CISCO and send a BYE immediately or
Asterisk can detect a hangup tone?

-- 

Best Regards
Charles
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Re: [Asterisk-Users] Registration to multiple GKs

[Charles Wang]

Is it possible to run Asterisk with another GKs using Neighbor mode? 
If it is possible, we can run asterisk with several gnugks. 

On Apr 2, 2005 10:41 PM, Alex Vishnev <[EMAIL PROTECTED]> wrote:
> I don't think you can. The rules of h323 is so that you can register with a
> single gk at a time.
> 
> Alex
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie
> Sent: Saturday, April 02, 2005 6:37 AM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] Registration to multiple GKs
> 
> Hi all,
> 
> How can I configure chan_h323 or chan_oh323 to register to multiple GK
> and route calls in-between?
> 
> Many thanks.
> Newbie
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Charles
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[Asterisk-Users] HELP: How to configure h323 channel driver ?

[Charles Wang]

Hi, ALL:

I has installed my chan_h323 channel driver in my *.
my scenario is:

SIP UA => SER(mediaproxy) => Asterisk => chan_h323 => GNUGK => H323 EP
And my UA and EP all support codecs such as alaw ulaw & G.729 at least.
I dial from UA behind NAT to H323 EP, and I answer from H323 EP too.
But I can not hear any voice from each side. Can anybody point out why it is?

h323.conf
--
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
accountcode = myaccountname
gatekeeper = IP of GNUGK
AllowGKRouted = yes
amaflags=default
type=h323
prefix=888248
e164=8881238
context=voip323
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw
allow=g723.1


extensions.conf
--
[general]
static=yes
writeprotect=no

[globals]

[default]
exten => _.,1,Dial(H323/${EXTEN})



-- 

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Charles
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Re: [Asterisk-Users] H323 <=> SIP Converter for Asterisk compertable

[Charles Wang]

Hi, ALL:

I'm almost give up the oh323 too.
I compiled the asterisk-oh323 for several times(ten or more). 
In my first time, I got pwlib 1.6.6 from CVS of openh323.org. 
But it seems a little buggy,so I failed.

I downloaded Janus's patch version, and followed its steps. It seems
OK when I compile with pwlib 1.6.6 and openh323 1.13.5 plus oh323
v0.7.1.

But I want my Asterisk to transfer my SIP call to H323 gatekeeper.
  SIP UAC ==> SER ==> Asterisk ==> GNUGK =X=> some H323 EPs.

There are too few documents or mailling lists to tell me how to configure it.

I want my asterisk registering to GNUGK as a gateway mode. 
I know how to set it up on my GNUGK's gatekeeper.ini. 
But does anyone kind to tell me how to configure my extensions.conf
and oh323.conf?

I can make a call to reach GNUGK. But the call will be hangup for some
reasons(I don't know what reason it is) when I find ACF and
unconnected CDR on my GNUGK's log. There is NOT any EP rings at that
time.

It is very difficult to setup such a environment and too few users
discuss about it.
Is the way just give it up?



On Wed, 23 Mar 2005 19:33:55 +0100, Yves <[EMAIL PROTECTED]> wrote:
> Try to isolate the problems, and send bugs to :
> https://skylab.inaccessnetworks.com/mantis/main_page.php
> 
> Doing this will improve the project.
> 
> We're using it and it's working pretty good.
> 
> Don't give up too fast!
> 
> Yves
> 
> 
> Bashir Ullah - www.Lamsre.Com wrote:
> > Hi George
> >
> > I did install and checkup several times, but some times h323 gateway or
> > softswitch cant accept my call and was able to accept call but no sound. so
> > can you help me please to implement a h323 solution. You may contact with me
> > if you want.
> >
> > Thanks
> >
> > Bashir
> > Call. 1-604 323 7991
> > Mail. [EMAIL PROTECTED]
> >
> >
> >
> > - Original Message -
> > From: "George K. Konstantoulakis" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Wednesday, March 23, 2005 3:11 AM
> > Subject: Re: [Asterisk-Users] H323 <=> SIP Converter for Asterisk
> > compertable
> >
> >
> >
> >>Hello Bashir,
> >>
> >>what kind of problems are you having with oh323 ?
> >>
> >>George
> >>
> >>Bashir Ullah - www.Lamsre.Com wrote:
> >>
> >>
> >>>Hi All * lover.
> >>>
> >>>This is not a question only this is a request to all SIP and Asterisk
> >
> > user .
> >
> >>>I am also with asterisk last few month and providing callingcard
> >
> > soluation.
> >
> >>>most of the SIP or IAX provider asking very high price which is really
> >
> > tough
> >
> >>>to resell in market. but still there is some h323 provider offering good
> >>>price. so as a asterisk user i tried so many times and now give up to
> >>>implement oh323, h323 by asterisk. i am sorry and also there is very may
> >
> > be
> >
> >>>none user for asterisk with h323. Thats why need a seperate soluation and
> >>>open source for converter h323 to sip vies-versa for asterisk user.
> >>>
> >>>Is it possible in near future. or is there any solution already done with
> >
> > is
> >
> >>>open source.
> >>>
> >>>
> >>>Thanks for your time to read this mail.
> >>>
> >>>Bashir
> >>>
> >>>___
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> >>>
> >>>
> >>
> >>___
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> >
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Re: [Asterisk-Users] Re: how to sip->h323 using asterisk-oh323-0.7.1

[Charles Wang]

I also have problems the same with you.
I can find my asterisk registered on my GK's status port(7000).
And I make a call from my XPro to SIP and SIP to Asterisk, then
Asterisk calls to a H323 phone via GNUGK.

I can find the CDR message on GK's status monitor. But I the GK only
first ACF(tail with false) and unconnected CDR on my GK.

Do you solve your problem? Can you share me your asterisk config such
as extensions.conf and oh323.conf, gatekeeper.ini for me to refer? 
Please.

Best Regards
Charles


On Thu, 10 Mar 2005 23:33:53 -0800 (PST), Kamran Ahmad <[EMAIL PROTECTED]> 
wrote:
> hello
> 
> i am using my own gnugatekeeper as a gatekeeper for my
> asterisk. asterisk is registering successfully with
> Gnugatekeeper. but it is not transfering call to
> gnugk.
> 
> any one guide me who to do this
> --
> SJPhone(sipSoftPhone using sip)->asterisk
> asterisk(conversion from sip -> h.323)
> asterisk(send h.323)->GnuGK
> GnuGk->SoftPhone(h.323 OpenPhone)
> -
> 
> on GnuGatekeeper side
> gatekeeper.ini
> 
> [Gatekeeper::Main]
> Fourtytwo=42
> TimeToLive=600
> 
> [RoutedMode]
> GKRouted=1
> H245Routed=0
> CallSignalPort=1721
> 
> [RasSrv::PermanentEndpoints]
> 192.168.0.203=xyz;123
> 
> [GkStatus::Auth]
> rule=allow
> 
> on asterisk
> oh323.conf
> ---
> ;
> ; Configuration file of OpenH323 channel driver
> ;
> 
> ;-
> ; General configuration options
> ; (ports, jitter, GK, ...)
> ;-
> [general]
> listenAddress=192.168.0.203
> listenPort=1719
> connectPort=1719
> 
> tcpStart=1
> tcpEnd=2
> 
> udpStart=1
> udpEnd=2
> 
> fastStart=yes
> 
> h245Tunnelling=no
> 
> h245inSetup=no
> 
> inBandDTMF=yes
> 
> silenceSuppression=no
> 
> jitterMin=20
> jitterMax=100
> 
> ipTos=none
> tos=lowdelay
> outboundMax=10
> inboundMax=10
> simultaneousMax=10
> 
> wrapLibTraceLevel=1
> libTraceLevel=1
> libTraceFile=stdout
> 
> gatekeeper=192.168.0.153
> gatekeeperPassword=test1
> accountcode=test1
> gatekeeperTTL=600
> 
> userInputMode=TONE
> 
> amaFlags=default
> 
> context=default
> 
> [xyz]
> type=h323
> prefix=123
> context=default
> 
> alias=1234
> context=default
> ;-
> ; Specify and configure CODEC related
> ; options
> ;-
> [codecs]
> codec=G711U
> frames=20
> 
> extensions.conf
> --
> [default]
> exten=>2000,1,Dial(SIP/${EXTEN})
> exten=>3000,1,Dial(SIP/${EXTEN})
> exten=>_123,1,Dial(SIP/${EXTEN})
> exten=>_321,1,Dial(OH323:h323/[EMAIL PROTECTED]:1719|30|r)
> 
> sip.conf
> --
> [2000]
> host=dynamic
> type=friend
> dtmfmode=INFO
> canreinvite=no
> 
> [3000]
> host=dynamic
> type=friend
> dtmfmode=INFO
> canreinvite=no
> 
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-- 

Best Regards
Charles
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[Asterisk-Users] HELP: Failed start after install asterisk_oh323-0.7.1

[Charles Wang]

Hi, ALL:

I install my oh323 channel driver following steps of 
http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en

I works my asterisk well before install the chan_oh323.so. But after I
do "make install" the oh_323, my asterisk crash and show me the
following message (asterisk -vvc).

Does anyone have any idea about it? What's wrong about ir?

-- Error Message --
 [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver)
Mar 21 11:13:25 WARNING[16199]: config_old.c:27 ast_load: ast_load is
deprecated, use ast_config_load instead!
  == Parsing '/etc/asterisk/rtp.conf': Found
Mar 21 11:13:25 WARNING[16199]: config_old.c:39 ast_destroy:
ast_destroy is deprecated, use ast_config_destroy instead!
  == Parsing '/etc/asterisk/oh323.conf': Found
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
[1]WrapGatekeeperServer::WrapGatekeeperServer: Creating new gatekeeper.
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault

 oh323.conf 
[general]
listenAddress=myip
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=5
libTraceFile=/var/log/asterisk/oh323.log
gatekeeper=mygnugk
;gatekeeperPassword=secret
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=myaccount
context=voip-h323

[register]
alias=h323248
alias=248

[codecs]
codec=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729
;frames=2


-- 

Best Regards
Charles
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[Asterisk-Users] HELP: Dose G.729 with IPP only worked on Intel CPU?

[Charles Wang]

Hi, ALL:

I install IPP(l_ipp_ia32_itanium_p_4_1_2.tar) and download the speech codeing
(l_ipp-sample-speech-coding_p_4.1.008.tgz) then patch it (g729-041103.diff).

My CPU is Centaur VIA Nehemiah with 998.715 MHz processor not INTEL CPU.

I choose PIII as its CPU type when I modify Makefile under "G729-float".

# For PIII 
OPTIMIZE= -O6 -mcpu=pentium3 -march=pentium3 -ffast-math -fomit-frame-pointer 
IPPCORE=a6 

I got the codec_g729.so and copy it to /usr/lib/asterisk/modules/.
Modified /etc/init.d/asterisk and add LD_LIBRARY_PATH and export it.

Modified /etc/asterisk/sip.conf and add allow=g729.

I worked my asterisk well before add G.729 codec. But after it, my asterisk 
crashed a few seconds after I run a startup command
"/etc/init.d/asterisk start".

Does anyone have the same problem?



-- 

Best Regards
Charles
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Re: [Asterisk-Users] ASTCC - how to use different brands?

[Charles Wang]

I had installed it and found out only the field of inc in your account table.
For example, you have A and B brands. A's inc is 6, B's inc is 60.
When you create a user belong to A brand. It will use 6 seconds as its
includedseconds.

Best Regards
Charles


On Tue, 15 Mar 2005 23:34:38 +0800, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> Nabeel Jafferali wrote:
> 
> >>I just downloaded the new astcc and it includes now a new
> >>field in the list of the cards: Brand Great!
> >>How can I use it in the dialplan?
> >>
> >>
> >
> >You can't use it in the dialplan, you use it when creating a card.
> >
> >
> >
> 
> HOW do I use it?
> 
> I could not find anything useful with BRAND, if it just a word!!!
> 
> I would need that a user could choose between two tarriffs, ... I
> thought that would be great to use Brands for that.
> 
> bye
> 
> Ronald
> 
> --
> Ronald Wiplinger  (CEO of ELMIT)
> http://www.elmit.com+886 (0) 939--77-55-16  or FWD 511208
> - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org
> 
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[Asterisk-Users] What different between asterisk-oh323 and astersk's chan_h323 ?

[Charles Wang]

Dear ALL:

I find that everybody discuss with asterisk-oh323 instead of chan_h323
of asterisk channels. Why do they choose asterisk-oh323 and not use
the build-in module (chan_h323)? What different between these two
componments?

In my guess, the build-in module should be easy to implement rather
than setup another application(asterisk-oh323) expect some bugs or
failed functions on chan_h323.

Dose anybody tell me how to implement a Asterisk to H323(register to
gatekeeper)?

I want my asterisk(as GW mode) register to GNUGK.

Best Regards 
Charles
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[Asterisk-Users] Where to download the asterisk-oh323?

[Charles Wang]

Dear ALL:

Where can I find the oh323 module on CVS or anywhere?

I want to implement the SIP(ser) to Asterisk to H323(gnugk).

Thank you.

Best Regards 
Charles
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Re: [Asterisk-Users] Re: how to sip->h323 using asterisk-oh323-0.7.1

[Charles Wang]

Dear Kamran:

Could you also post your gatekeeper.ini about connect to asterisk and
what version of GNUGK you used?


On Wed, 9 Mar 2005 06:44:54 -0800 (PST), Kamran Ahmad <[EMAIL PROTECTED]> wrote:
> 
> 
> i am using gnugatekeeper. i have three things
> gatekeeper ip, account, accountpassword how to set
> account and password in oh323.conf
> 
> gatekeeper=gnu gatekeeper ip
> gatekeeperPassword=accountpassword
> accountCode=account
> 
> is this ok any example how to use this i want to rout
> my sip call to this gatekeeper for h323.
> 
> 
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Re: [Asterisk-Users] Asterisk + SER

[Charles Wang]

Yes, I use this method too.


On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak <[EMAIL PROTECTED]> wrote:
> you do not need radius for ser and asterisk to speak to each other. if
> anything, i would suggest using SER for the endpoint and asterisk for
> the billing and accounting.
> 
> -yair
> 
> 
> On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford <[EMAIL PROTECTED]> wrote:
> > I just installed SER last night but if you want it ot talk to Asterisk I
> > found that you should install FREERADIUS Server and RADIUS CLIENT. For it to
> > function properly
> >
> > - Original Message -
> > From: "Nitesh Divecha" <[EMAIL PROTECTED]>
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > 
> > Sent: Friday, February 25, 2005 8:29 PM
> > Subject: [Asterisk-Users] Asterisk + SER
> >
> > > Hello All,
> > >
> > > Has anyone tried Asterisk with SER.?
> > > My main focus is billing and authentication of my endpoints.
> > >
> > > I want Asterisk to handle all my endpoints and SER to do
> > > billing/accounting
> > > stuff.
> > >
> > > Any help will be highly appreciated.
> > >
> > > Neel
> > >
> > >
> > >
> > > ___
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Re: [Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got an error

[Charles Wang]

Dear Martijn:

Thank you for your tips. It is very important for me such an asterisk newbie. 

I have recompile the source of asterisk-1.0.5 and compile the
asterisk-addon done.

Anyway, thank you for your kind.

Best Regard
Charles


On Sat, 26 Feb 2005 16:25:15 +0100, Martijn van Oosterhout
<[EMAIL PROTECTED]> wrote:
> See bug #3639
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0003639
> 
> Nothing's been committed yet though I think...
> 
> On Sat, Feb 26, 2005 at 10:58:11PM +0800, Charles Wang wrote:
> > Dear ALL:
> >
> > I got an error message lists below.
> >
> > Does anyone have the same problem? How to solve it?
> >
> > Best Regard
> > Charles
> >
> > In file included from config.c:34:
> > include/asterisk/app.h:62: array size missing in `options'
> > make: *** [config.o] Error 1
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> Ecomtel Pty Ltd
>
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[Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got an error

[Charles Wang]

Dear ALL:

I got an error message lists below.

Does anyone have the same problem? How to solve it?

Best Regard
Charles

In file included from config.c:34:
include/asterisk/app.h:62: array size missing in `options'
make: *** [config.o] Error 1
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[Asterisk-Users] ERROR: compile asterisk (from CVS HEAD)

[Charles Wang]

Dear ALL:

I got an error message lists below.

Does anyone have the same problem? How to solve it?

Best Regard
Charles

In file included from config.c:34:
include/asterisk/app.h:62: array size missing in `options'
make: *** [config.o] Error 1
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[Asterisk-Users] ERROR: when compile app_addon_sql_mysql.c of asterisk_addon

[Charles Wang]

Dear ALL:

I got this error when try to compile asterisk_addon.

Does anybody have solution about it ?

cc -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4
arguments, but only 3 given
make: *** [app_addon_sql_mysql.o] Error 1

Best Regard
Charles
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[Asterisk-Users] Question of SER to Asterisk to PSTN

[Charles Wang]

Dear ALL:

My scenario lists below:

Assume: UA1 with sip id "1011"
  And dial number to PSTN is "0939749xxx"
  There is no modification rule at my CISCO.
  (It will not change any dialed number)

UA1 ==> SER ==> UA2 
(SIP to SIP)
UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==> PSTN  (SIP to PSTN)
 port:5060  port:5065port:5060
 IP:xxx.xxx.190.248IP:xxx.xxx.190.243
(On the same server)  (On another server)


I know how to forward a call from ser to CISCO 5300. And I have done it ever.
UA1 ==> SER ==> CISCO 5300 ==> PSTN

Now I modify the ser.cfg, and want to forward all calls to PSTN to the
port 5065 of Asterisk on the same server.
And Asterisk should accept the call and forward it to PSTN via CISCO 5300.

1. UA1's SER SIP id is "1011"
  (Assume all SIP account is 4 digists and first leader number is not zero)
2. UA1 want to dial my mobile phone number  0939749xxx
3. CISCO will forward this call to PSTN,

I only modify extensions.conf & sip.conf.

I can make a call from UA1 to 0939749xxx via SER and Asterisk to CISCO.
But I can not hear any voice in UA1 or my mobile phone.

What's wrong in my setting? Does Anybody help me?

Thank you for your kind to read and answer my question.

PS: How can I set the maxinum log level to /var/log/asterisk/messages ? 
  I start my asterisk with the sample of redhat and
"/etc/init.d/asterisk restart".



Best Regard
Charles


In my extensions.conf:
---
[general]
static=yes
writeprotect=no

[globals]
SERADDRESS=xxx.xxx.190.248:5060
CISCOADDRESS=xxx.xxx.190.243:5060

[ser]
exten => _XX,1,Dial(SIP/[EMAIL PROTECTED],60)

[cisco5300]
exten => 1055,1,Dial(SIP/[EMAIL PROTECTED],60)

In my sip.conf:
---
[general]
context=ser 
port=5065
bindaddr=0.0.0.0
register => 1055:[EMAIL PROTECTED]   ; using to register to SER SIP server

[cisco5300_pstn] 
type=friend 
defaultip=xxx.xxx.190.243   
amaflags=default   ; Choices are default, omit, billing,
documentation
disallow=all 
allow=ulaw 
allow=alaw 
allow=g729 
allow=g723.1
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[Asterisk-Users] Do ser + asterisk_b2bua work ?

[Charles Wang]

Dear ALL:

I find a program named "asterisk_b2bua" on
http://developer.berlios.de/projects/b2bua/

And I also download them(two components) and try to test it.

But I have not enough knowledge about asterisk. It seems a Software PBX.
Does asterisk_b2bua work? Does anybody ever try it?

I have questions about my scenario.

  |==> UA2  (Internet)
  |
UA1 ===> SER ===> Asterisk B2BUA ===> Trunking A (PSTN)
|
|
 CDR + Prepaid + Handle Calls(Tear-down when call
during limited)
|
|
  Authentication ( Radius / DB )

Q1. Will Asterisk's B2BUA pass through the "To" (such as
 [EMAIL PROTECTED] PSTN number) to Trunking A?
 In another word, is the B2BUA not necessary to rewrite the phone number?

Q2. How can B2BUA know when to tear-down this call if the call has some limits.
 For example, UA1 has only 120 seconds to use this International call.

Q3. Is it necessary for Radius Authentication? Is it possible if no
radius exists
 and SER and Asterisk use the same MySQL database.

Q4. What do I need to download and setup when I use Asterisk
 to start this prepaid compoment? The first one shall be
asterisk-1.0.5.tar.gz.
 Maybe some Mysql supported scripts or installaion. And any more??

Best Regard
Charles
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