Re: [asterisk-users] Asterisk Query
2010/4/29 garge rama
>
>
> Hi,
>
>
>
> I am new to asterisk and trying to make calls with TDM400P asterisk digium
> card.
>
>
>
> I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
> libpri-1.4.10.2 packages which are downloaded from asterisk website (
> www.asterisk.org)
>
> and able to compile successfully. TDM400P Digium card (having only one FXS
> connected to J4) has installed successfully in PC.
>
>
>
> I would like to make calls across SIP [x-lite] to analog phone connected to
> TDM400P Digium card (fxs-j4).
>
> For this the following four conf files are modified as shown below.
>
>
>
> * chan_dahdi.conf*
>
> *==*
>
> [channels]
>
> context=test
>
> usecallerid=yes
>
> hidecallerid=no
>
> immediate=no
>
>
>
> signaling=fxo_ks
>
> echocancel=yes
>
> group=1
>
> channel=1
>
>
>
> *extensions.conf***
>
> *=*
>
> [my-phones] --->*EXTEN does not exists for your sip
> peer context*
>
> exten => 2000,1,Dial(SIP/2000)
>
> ; Should look like:
>
*exten => ,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you
want})
> [test]
>
> exten => ,1,Dial(Zap/1)
>
> exten => ,2,HangUp()
>
>
>
> *sip.conf***
>
> *===*
>
> [general]
>
> port = 5060
>
> bindaddr = 0.0.0.0
>
> context = others
>
>
>
> [2000]
>
> type=friend
>
> *context=**my-phones *
>
> secret=1234
>
> host=dynamic
>
>
>
> *system.conf*
>
> *==*
>
> fxoks=1
>
> loadzone = be
>
> defaultzone = be
>
>
>
> With those changes x-lite getting registered with asterisk and analog
> device/phone is getting ring tone with off-hook and also getting debug
> prints on cli, but not able to make calls.
>
>
>
> Test Setup:
>
>
>
> X-lite [configured as 2000, password… other info] running on asterisk PC
> à registered with asterisk.
>
> Analog phone connected to TDM400P Digium card - FXS-J4 running on same
> asterisk PC à getting ring tone
>
>
>
> Test Result:
>
> =
>
> Tried by calling from x-lite à getting message on CLI “call from
> ‘2000’ to ‘’ rejected because extension not found”
>
> Tried by calling 2000 from analog phone [Digium-FXS-J4] -> getting some
> engage/disconnected tone while pressing digts [2000] on phone itself.
>
>
>
> Welcome for your valuable suggestions and comments. Thank You in advance.
>
>
>
> Regards,
>
> Garge.
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
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>
--
Juan.
Linux User #441131
--
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Re: [asterisk-users] Asterisk Query
Hi Juan,
Thanks for your inputs, I tried with changes you suggested and find my
observation.
After adding context and extension able to make an outgoing call
[Digium-fxs<> to X-lite<2000>].
But not able to make incoming call [X-lite<2000> to Digium-fxs<>]. Call
failed with,
(1) “*Call failed: 503 Service Unavailat *” error message on X-lite
(2) “CHANUNAVAIL” on asterisk CLI.
**CLI>> Saved useragent "X-Lite release 1105d" for peer 2000*
* == Using SIP RTP CoS mark 5*
*-- Executing [3...@my-phones:1] Dial("SIP/2000-", "Zap/1/")
in new stack*
*[May 6 13:02:44] WARNING[20496]: channel.c:4003 ast_request: No channel
type registered for 'Zap'*
*[May 6 13:02:44] WARNING[20496]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'Zap' (cause 66 - Channel not implemented)*
* == Everyone is busy/congested at this time (1:0/0/1)*
*-- Auto fallthrough, channel 'SIP/2000-' status is
'CHANUNAVAIL'*
Please find conf files below.
chan_dahdi.conf
[channels]
context=my-phones
usecallerid=yes
hidecallerid=no
immediate=no
signaling=fxo_ks
echocancel=yes
group=1
channel=1
sip.conf
==
[general]
port=5060
bindaddr=0.0.0.0
context=my-phones
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
extensions.conf
===
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => ,1,Dial(Zap/1/)
system.conf
fxoks=1
loadzone=us
defaultzone=us
Please let me know any other configuration needs to be done.
On Fri, Apr 30, 2010 at 1:12 AM, Juan David Diaz wrote:
>
>
> 2010/4/29 garge rama
>
>>
>>
>> Hi,
>>
>>
>>
>> I am new to asterisk and trying to make calls with TDM400P asterisk digium
>> card.
>>
>>
>>
>> I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
>> libpri-1.4.10.2 packages which are downloaded from asterisk website (
>> www.asterisk.org)
>>
>> and able to compile successfully. TDM400P Digium card (having only one FXS
>> connected to J4) has installed successfully in PC.
>>
>>
>>
>> I would like to make calls across SIP [x-lite] to analog phone connected
>> to TDM400P Digium card (fxs-j4).
>>
>> For this the following four conf files are modified as shown below.
>>
>>
>>
>> * chan_dahdi.conf*
>>
>> *==*
>>
>> [channels]
>>
>> context=test
>>
>> usecallerid=yes
>>
>> hidecallerid=no
>>
>> immediate=no
>>
>>
>>
>> signaling=fxo_ks
>>
>> echocancel=yes
>>
>> group=1
>>
>> channel=1
>>
>>
>>
>> *extensions.conf***
>>
>> *=*
>>
>> [my-phones] --->*EXTEN does not exists for your
>> sip peer context*
>>
>> exten => 2000,1,Dial(SIP/2000)
>>
>> ; Should look like:
>>
> *exten => ,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you
> want})
>
>> [test]
>>
>> exten => ,1,Dial(Zap/1)
>>
>> exten => ,2,HangUp()
>>
>>
>>
>> *sip.conf***
>>
>> *===*
>>
>> [general]
>>
>> port = 5060
>>
>> bindaddr = 0.0.0.0
>>
>> context = others
>>
>>
>>
>> [2000]
>>
>> type=friend
>>
>> *context=**my-phones *
>>
>> secret=1234
>>
>> host=dynamic
>>
>>
>>
>> *system.conf*
>>
>> *==*
>>
>> fxoks=1
>>
>> loadzone = be
>>
>> defaultzone = be
>>
>>
>>
>> With those changes x-lite getting registered with asterisk and analog
>> device/phone is getting ring tone with off-hook and also getting debug
>> prints on cli, but not able to make calls.
>>
>>
>>
>> Test Setup:
>>
>>
>>
>> X-lite [configured as 2000, password… other info] running on asterisk PC
>> à registered with asterisk.
>>
>> Analog phone connected to TDM400P Digium card - FXS-J4 running on same
>> asterisk PC à getting ring tone
>>
>>
>>
>> Test Result:
>>
>> =
>>
>> Tried by calling from x-lite à getting message on CLI “call from
>> ‘2000’ to ‘’ rejected because extension not found”
>>
>> Tried by calling 2000 from analog phone [Digium-FXS-J4] -> getting some
>> engage/disconnected tone while pressing digts [2000] on phone itself.
>>
>>
>>
>> Welcome for your valuable suggestions and comments. Thank You in advance.
>>
>>
>>
>> Regards,
>>
>> Garge.
>>
>>
>>
>> --
>>
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Juan.
> Linux User #441131
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-u
Re: [asterisk-users] Asterisk Query
Hi Garge -
>> exten =>
>> ,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you want})
Two things:
1. There is no such thing as Zap anymore. Zap has been renamed to
Dahdi because of a trademark issue. So your extension should look
like:
exten = ,Dial(Dahdi/1/)
2. Do you really mean to dial ''? This number should be a valid
phone number.
- Noah
--
_
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Re: [asterisk-users] Asterisk Query
Hi Noah,
Thank You.After changing Zap to Dahdi in conf files, able to make calls Now.
Regards,
Garge.
On Thu, May 6, 2010 at 8:56 PM, Noah Miller wrote:
> Hi Garge -
>
> >> exten =>
> >> ,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you
> want})
>
> Two things:
>
> 1. There is no such thing as Zap anymore. Zap has been renamed to
> Dahdi because of a trademark issue. So your extension should look
> like:
>
> exten = ,Dial(Dahdi/1/)
>
> 2. Do you really mean to dial ''? This number should be a valid
> phone number.
>
>
> - Noah
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] asterisk query mysql problem or bug?
Don't use commas as delimiters in database. You must use pipe |. Replace
your commas and see if that does the trick.
-Matthew
Wei Kun wrote:
Hi;
I have entries as below in DB,
mysql> select * from sip_buddies;
++--+--++-+++---
-++--+--+
| id | name | context | defaultip | host| mailbox| type |
regseconds | ipaddr | username | port |
++--+--++-+++---
-++--+--+
| 1 | 2000 | from-sip | 10.1.2.192 | dynamic | [EMAIL PROTECTED] | friend |
1123733887 | 10.1.2.192 | 2000 | 5060 |
| 2 | 2001 | from-sip | 10.1.2.220 | dynamic | [EMAIL PROTECTED] | friend |
1123733888 | 10.1.1.220 | 2001 | 5080 |
++--+--++-+++---
-++--+--+
2 rows in set (0.01 sec)
mysql> select * from extensions_table;
++--+---+--+---++
| id | context | exten | priority | app | appdata|
++--+---+--+---++
| 1 | from-sip | 2000 |1 | Dial | SIP/2000,20|
| 2 | from-sip | 2000 |2 | Voicemail | u2000 |
| 3 | from-sip | 2000 | 102 | Voicemail | b2000 |
| 4 | from-sip | 2000 | 103 | Hangup||
| 5 | from-sip | 2001 |1 | Dial | SIP/2001 |
| 6 | from-sip | 2001 |2 | Voicemail | u2001 |
| 7 | from-sip | 2001 | 102 | Voicemail | b2001 |
| 8 | from-sip | 2001 | 103 | Hangup||
| 9 | from-sip | 2999 |1 | VoicemailMain | ${CALLERIDNUM} |
++--+---+--+---++
9 rows in set (0.00 sec)
Somehow the program get the info '2001,20' stripped from extensions_table
appdata column 'SIP/2001, 20', and try to look it up in sip_buddies name
column as debug output below.
Aug 11 12:23:05 DEBUG[23952] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sip_buddies WHERE name = '2001,20'
Of course, it can't find it, and go to second step for voicemail. If I
change the appdata to 'SIP/2001', it can find it and ring remote party, the
problem is it rings for ever without the 20 hint.
Any hints for this problem?
Thanks
Kun
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RE: [Asterisk-Users] asterisk query mysql problem or bug?
It does the trick!
Thanks
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Boehm
Sent: Thursday, August 11, 2005 11:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk query mysql problem or bug?
Don't use commas as delimiters in database. You must use pipe |. Replace
your commas and see if that does the trick.
-Matthew
Wei Kun wrote:
> Hi;
> I have entries as below in DB,
>
> mysql> select * from sip_buddies;
>
++--+--++-+++---
> -++--+--+
> | id | name | context | defaultip | host| mailbox| type |
> regseconds | ipaddr | username | port |
>
++--+--++-+++---
> -++--+--+
> | 1 | 2000 | from-sip | 10.1.2.192 | dynamic | [EMAIL PROTECTED] | friend |
> 1123733887 | 10.1.2.192 | 2000 | 5060 |
> | 2 | 2001 | from-sip | 10.1.2.220 | dynamic | [EMAIL PROTECTED] | friend |
> 1123733888 | 10.1.1.220 | 2001 | 5080 |
>
++--+--++-+++---
> -++--+--+
> 2 rows in set (0.01 sec)
>
> mysql> select * from extensions_table;
> ++--+---+--+---++
> | id | context | exten | priority | app | appdata|
> ++--+---+--+---++
> | 1 | from-sip | 2000 |1 | Dial | SIP/2000,20|
> | 2 | from-sip | 2000 |2 | Voicemail | u2000 |
> | 3 | from-sip | 2000 | 102 | Voicemail | b2000 |
> | 4 | from-sip | 2000 | 103 | Hangup||
> | 5 | from-sip | 2001 |1 | Dial | SIP/2001 |
> | 6 | from-sip | 2001 |2 | Voicemail | u2001 |
> | 7 | from-sip | 2001 | 102 | Voicemail | b2001 |
> | 8 | from-sip | 2001 | 103 | Hangup||
> | 9 | from-sip | 2999 |1 | VoicemailMain | ${CALLERIDNUM} |
> ++--+---+--+---++
> 9 rows in set (0.00 sec)
>
> Somehow the program get the info '2001,20' stripped from extensions_table
> appdata column 'SIP/2001, 20', and try to look it up in sip_buddies name
> column as debug output below.
>
> Aug 11 12:23:05 DEBUG[23952] res_config_mysql.c: MySQL RealTime: Retrieve
> SQL: SELECT * FROM sip_buddies WHERE name = '2001,20'
>
> Of course, it can't find it, and go to second step for voicemail. If I
> change the appdata to 'SIP/2001', it can find it and ring remote party,
the
> problem is it rings for ever without the 20 hint.
>
> Any hints for this problem?
>
> Thanks
> Kun
>
> ___
> Asterisk-Users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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