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 noahisaacmil...@gmail.comwrote:
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 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-lite2000].
But not able to make incoming call [X-lite2000 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 juanch...@gmail.comwrote:
2010/4/29 garge rama garge.r...@gmail.com
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.
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Linux User #441131
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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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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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[asterisk-users] Asterisk Query
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 = 2000,1,Dial(SIP/2000) [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 àregistered 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
Re: [asterisk-users] Asterisk Query
2010/4/29 garge rama garge.r...@gmail.com
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 --
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asterisk-users mailing list
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--
Juan.
Linux User #441131
--
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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
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[Asterisk-Users] asterisk query mysql problem or bug?
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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