[asterisk-users] help please (D 300 JCT)
i need to know how can i configure a D 300JCT with asterisk, i want to connect two PBX where each one have this card on it,i really need your help as soon as possible. i already done some file configuration system.conf and in chan_dahdi.conf and i have installed the DAHDI and the LibPri modules. -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-mail: harazta...@gmail.com ouabimedcha...@gmail.com * -- _ -- 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] Help Please - Asterisk MYSQL interface seems to be eating data
Al Baker schrieb: Are you saying the * server does NOT TRY to re-establish the BD connection ? The MySQL Realtime driver _does_ reconnect. (Search for mysql_reconnect() in res_config_mysql.c) If NOT, what happens to you CDR records ? Same thing with cdr_addon_mysql.c - it tries to reconnect. When there is no connection it writes the CDRs to a file and as soon as it successfully reconnects stores them in the database. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Steve Totaro schrieb: I would not run MySQL on the local box. I would simple use Asterisk's csv CDRs and then use some script to import the CSVs into a database residing on another server using some sort of script. Depending on your needs, you could probably run that during low call volume. I also think that you adapt the free queue_log to database script by Queuemetrics to do what you want on the fly. We're using a custom script for the queue_log - db import in Gemeinschaft as well. But I'm not really happy with that. You need to run such a script at least once a minute to get real-time statistics for the GUI etc. Everything could be much nicer if Asterisk wrote the queue log into the database directly. As an alternative solution you can use a named pipe. But Asterisk is not prepared to handle the broken pipe error which occurs if your script should ever fail to read from the pipe. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Are you saying the * server does NOT TRY to re-establish the BD connection ? Does your whole * SERVER freeze ? If NOT, what happens to you CDR records ? Anthony Francis wrote: Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Tue, May 6, 2008 at 11:42 AM, Anthony Francis [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP I would not run MySQL on the local box. I would simple use Asterisk's csv CDRs and then use some script to import the CSVs into a database residing on another server using some sort of script. Depending on your needs, you could probably run that during low call volume. I also think that you adapt the free queue_log to database script by Queuemetrics to do what you want on the fly. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG (or a
brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is eaten
ENTIRELY by Asterisk. If several fields are requested, the First Field Is
Eaten and the remaining filed are returned, but place in the WRONG Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)
exten = ,n,NoOp(CUSTID is ${custid})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
Here is the Asterisk CLI Output
dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id from agent where phone='1234567890') in new stack
-- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6 custid)
in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT ID is
.. 6 CONNID is 5) in new stack
-- Executing [EMAIL PROTECTED]:6] MYSQL(OSS/dsp, Clear 6) in new stack
-- Executing [EMAIL PROTECTED]:7] MYSQL(OSS/dsp, Disconnect 5) in new
stack
-- Executing [EMAIL PROTECTED]:8] Hangup(OSS/dsp, ) in new stack
== Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'
Hangup on console
*CLI dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id, status, tlce from agent where phone='1234567890') in new stack
-- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6 custid
mystatus mytlce) in new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is free MYSTATUS is
2008-04-17 02:32:02 MYTLCE is ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT ID is
.. 6 CONNID is 5) in new stack
-- Executing [EMAIL PROTECTED]:6] MYSQL(OSS/dsp, Clear 6) in new stack
-- Executing [EMAIL PROTECTED]:7] MYSQL(OSS/dsp, Disconnect 5) in new
stack
--
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
I looked all over VOIP-INFO and ATFOT and could not find anything that
said or even suggested not using the mysql driver.(except NOT to have
BOTH drivers loaded at the same time). I could easily be missing
something. But the apparent BUG I am seeing is at such a Basic and
Simple Level of functionality that either DIGIUM ought to fix it ASAP or
update VOIP-INFO pages and their own documentation to say Broke - No
Workie and We Are No Gonna Fixie :)
Steve Totaro wrote:
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG (or a
brainfart on my part) with the MySQL add-on for Asterisk this is of FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is eaten
ENTIRELY by Asterisk. If several fields are requested, the First Field Is
Eaten and the remaining filed are returned, but place in the WRONG Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)
exten = ,n,NoOp(CUSTID is ${custid})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
Here is the Asterisk CLI Output
dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id from agent where phone='1234567890') in new stack
-- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6
custid) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT ID
is
.. 6 CONNID is 5) in new stack
-- Executing [EMAIL PROTECTED]:6] MYSQL(OSS/dsp, Clear 6) in new stack
-- Executing [EMAIL PROTECTED]:7] MYSQL(OSS/dsp, Disconnect 5) in new
stack
-- Executing [EMAIL PROTECTED]:8] Hangup(OSS/dsp, ) in new stack
== Spawn extension (default, , 8) exited non-zero on 'OSS/dsp'
Hangup on console
*CLI dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5 SELECT
cust_id, status, tlce from agent where phone='1234567890') in new stack
-- Executing [EMAIL
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
User intervention is required Al. You need to open a bug report on
mantis and anyone can edit the wiki (this is the nature of a wiki).
http://bugs.digium.com/main_page.php
Anyways, I am fairly certain that the UnixODBC connector is the
preferred way of connecting to MySQL.
Thanks,
Steve Totaro
On Mon, May 5, 2008 at 9:29 AM, Al Baker [EMAIL PROTECTED] wrote:
I looked all over VOIP-INFO and ATFOT and could not find anything that
said or even suggested not using the mysql driver.(except NOT to have
BOTH drivers loaded at the same time). I could easily be missing
something. But the apparent BUG I am seeing is at such a Basic and
Simple Level of functionality that either DIGIUM ought to fix it ASAP or
update VOIP-INFO pages and their own documentation to say Broke - No
Workie and We Are No Gonna Fixie :)
Steve Totaro wrote:
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG (or a
brainfart on my part) with the MySQL add-on for Asterisk this is of
FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is eaten
ENTIRELY by Asterisk. If several fields are requested, the First Field Is
Eaten and the remaining filed are returned, but place in the WRONG
Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE
is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus} MYTLCE
is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\ from\
agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)
exten = ,n,NoOp(CUSTID is ${custid})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is .. ${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
Here is the Asterisk CLI Output
dial
== Console is full duplex
*CLI -- Executing [EMAIL PROTECTED]:1] MYSQL(OSS/dsp, Connect connid
localhost ivr ivrxxx dtc) in new stack
-- Executing [EMAIL PROTECTED]:2] MYSQL(OSS/dsp, Query resultid 5
SELECT
cust_id from agent where phone='1234567890') in new stack
-- Executing [EMAIL PROTECTED]:3] MYSQL(OSS/dsp, Fetch fetchid 6
custid) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(OSS/dsp, CUSTID is ) in new
stack
-- Executing [EMAIL PROTECTED]:5] NoOp(OSS/dsp, FETCHID is 1 RESULUT
ID is
.. 6 CONNID is 5) in new stack
--
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
A quote from Tilghman Lesher from a previous post.
That's fine, but I have had the most horrid results using any distribution-
supplied ODBC drivers. The best results are obtained by source-compiling
the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
PsqlODBC. UnixODBC is fairly safe to use from distribution channels, however.
Thanks,
Steve Totaro
On Mon, May 5, 2008 at 10:06 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
User intervention is required Al. You need to open a bug report on
mantis and anyone can edit the wiki (this is the nature of a wiki).
http://bugs.digium.com/main_page.php
Anyways, I am fairly certain that the UnixODBC connector is the
preferred way of connecting to MySQL.
Thanks,
Steve Totaro
On Mon, May 5, 2008 at 9:29 AM, Al Baker [EMAIL PROTECTED] wrote:
I looked all over VOIP-INFO and ATFOT and could not find anything that
said or even suggested not using the mysql driver.(except NOT to have
BOTH drivers loaded at the same time). I could easily be missing
something. But the apparent BUG I am seeing is at such a Basic and
Simple Level of functionality that either DIGIUM ought to fix it ASAP or
update VOIP-INFO pages and their own documentation to say Broke - No
Workie and We Are No Gonna Fixie :)
Steve Totaro wrote:
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG (or
a
brainfart on my part) with the MySQL add-on for Asterisk this is of
FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is
eaten
ENTIRELY by Asterisk. If several fields are requested, the First Field
Is
Eaten and the remaining filed are returned, but place in the WRONG
Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have
tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly
show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus
mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus}
MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus
mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus}
MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\
from\
agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid)
exten = ,n,NoOp(CUSTID is ${custid})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. Any MySQL developers out there that can help us fix this? We need someone that has been developing towards the Mysql api. And please do not always refer to Digium developers when you have problems in Asterisk. There are developers that are not employees of Digium... Cheers, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Steve Totaro wrote:
A quote from Tilghman Lesher from a previous post.
That's fine, but I have had the most horrid results using any distribution-
supplied ODBC drivers. The best results are obtained by source-compiling
the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or
PsqlODBC. UnixODBC is fairly safe to use from distribution channels,
however.
And according to the O'Reilly book ODBC is the way to go.
Though they use PostgreSQL for their examples and Asterisk is installed
on a CentOS system the instructions are really good. Getting it to work
with MySQL should be pretty simple and I'm sure on-line resources for
doing this are be out there.
soapbox
Personally I never use MySQL except in cases where I am under extreme
duress. Therefore I tried and tossed trixbox, AsteriskNOW, and
freeePBX. Yes I know I can get around the database engine issue but
that is what a distribution should be for: no hacking (or at least
not-too-much) required.
It is now CentOS 5, Asterisk from source, PostgreSQL (on another system)
and hand edited (for now anyway) *.conf files. Maybe AsteriskGUI later.
/soapbox
Rod
--
Thanks,
Steve Totaro
On Mon, May 5, 2008 at 10:06 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
User intervention is required Al. You need to open a bug report on
mantis and anyone can edit the wiki (this is the nature of a wiki).
http://bugs.digium.com/main_page.php
Anyways, I am fairly certain that the UnixODBC connector is the
preferred way of connecting to MySQL.
Thanks,
Steve Totaro
On Mon, May 5, 2008 at 9:29 AM, Al Baker [EMAIL PROTECTED] wrote:
I looked all over VOIP-INFO and ATFOT and could not find anything that
said or even suggested not using the mysql driver.(except NOT to have
BOTH drivers loaded at the same time). I could easily be missing
something. But the apparent BUG I am seeing is at such a Basic and
Simple Level of functionality that either DIGIUM ought to fix it ASAP or
update VOIP-INFO pages and their own documentation to say Broke - No
Workie and We Are No Gonna Fixie :)
Steve Totaro wrote:
On Mon, May 5, 2008 at 4:21 AM, Al Baker [EMAIL PROTECTED] wrote:
I would appreciate any and all advice on what appears to be a BUG
(or a
brainfart on my part) with the MySQL add-on for Asterisk this is of
FEDORA 8
fully patched with Asterisk Addons 1-4-6 with the Asterisk 1.4.18.1
It appears that the interface eats the first field requested from a
table. If only One Field is Requested from the Table , that field is
eaten
ENTIRELY by Asterisk. If several fields are requested, the First
Field Is
Eaten and the remaining filed are returned, but place in the WRONG
Variable
since the 1tst fileld data was eaten. In the DIALPLAN below I have
tried 3
Different ways to approach this.
Extension – Get only ONE (1) field from Table
Extension – Get THREE(3) fields from the Table and Quote Them.
Extension - Get THREE(3) fields from the Table
I have show the Output from the Asterisk CL for each, which clearly
show
that SOMETHING is not
right. Maybe the Software, maybe the person using the software :)
Here is the Table in the Database.
mysql select * from agent;
+--+-+++-+
| id | cust_id | status |phone |tlce |
+--+-+++-+
| 0001 | NAMB | free | 1234567890 | 2008-04-17 02:32:02 |
| 0002 | NAMB | free | 2234567890 | 2008-04-17 02:32:02 |
| 0003 | NAMB | free | 3234567890 | 2008-04-17 02:32:02 |
| 0004 | NAMB | free | 4234567890 | 2008-04-17 02:32:02 |
+--+-+++-+
4 rows in set (0.00 sec)
Here is the DIALPLAN
exten = ,1,MYSQL(Connect connid localhost ivr ivrxxx dtc)
exten = ,n,MYSQL(Query resultid ${connid} SELECT\ cust_id\, \
status\,\ tlce\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus
mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS is ${mystatus}
MYTLCE is
${mytlce})
exten = ,n,NoOp(FETCHID is ${fetchid} RESULUT ID is ..
${resultid}
CONNID is ${connid})
exten = ,n,MYSQL(Clear ${resultid})
exten = ,n,MYSQL(Disconnect ${connid})
exten = ,n,HANGUP
exten = ,1,MYSQL(Query resultid ${connid} SELECT\ 'cust_id'\, \
'status'\,\ 'tlce'\ from\ agent\ where\ phone=\'1234567890\')
exten = ,n,MYSQL(Fetch fetchid ${resultid} custid mystatus
mytlce)
exten = ,n,NoOp(CUSTID is ${custid} MYSTATUS
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
I must be overlooking it, I pulled up the electronic version and searched for and read every instance where ODBC was mentioned and I could not find a single place where it said ODBC was to be the only or even the best method. If so I would never ever have gone down this road :( Quote And according to the O'Reilly book ODBC is the way to go. Roderick A. Anderson wrote: Steve Totaro wrote: A quote from Tilghman Lesher from a previous post. That's fine, but I have had the most horrid results using any distribution- supplied ODBC drivers. The best results are obtained by source-compiling the latest ODBC drivers, whether they be the MySQL ODBC Connector 3.51 or PsqlODBC. UnixODBC is fairly safe to use from distribution channels, however. And according to the O'Reilly book ODBC is the way to go. Though they use PostgreSQL for their examples and Asterisk is installed on a CentOS system the instructions are really good. Getting it to work with MySQL should be pretty simple and I'm sure on-line resources for doing this are be out there. soapbox Personally I never use MySQL except in cases where I am under extreme duress. Therefore I tried and tossed trixbox, AsteriskNOW, and freeePBX. Yes I know I can get around the database engine issue but that is what a distribution should be for: no hacking (or at least not-too-much) required. It is now CentOS 5, Asterisk from source, PostgreSQL (on another system) and hand edited (for now anyway) *.conf files. Maybe AsteriskGUI later. /soapbox Rod ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Monday 05 May 2008 10:16, Al Baker wrote: I must be overlooking it, I pulled up the electronic version and searched for and read every instance where ODBC was mentioned and I could not find a single place where it said ODBC was to be the only or even the best method. If so I would never ever have gone down this road :( While there are some things that you can do with the MYSQL app that you can't do with func_odbc, most of what people want to do (single row queries, updates, and inserts) can be done with func_odbc. Additionally, it's well supported, the author (me) is still around, and we're still considering enhancements. The MYSQL app is somewhat less certain. I don't know if the author is still around, it is largely in maintenance mode, and, due to a matter of licensing, the app is not in the Asterisk core distribution. ODBC is not going to be the only supported method of accessing a database; some native drivers exist, and they will continue to exist. The reason why ODBC is so important to us is that it allows us to interface with many varied databases out there for which the native API is not immediately available. Additionally, it significantly lightens our maintenance workload not to have to maintain drivers for some databases which our users may never use. And most of our database integration has been for the specific purpose of realtime. Our func_odbc module is about the most generic of database interfaces, and its usage is completely user-defined, unlike realtime or the other database interface (voicemail storage). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. Any MySQL developers out there that can help us fix this? We need someone that has been developing towards the Mysql api. And please do not always refer to Digium developers when you have problems in Asterisk. There are developers that are not employees of Digium... There used to be more outside developers, but Digium is a great place to work... ;-) -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? Any MySQL developers out there that can help us fix this? We need someone that has been developing towards the Mysql api. And please do not always refer to Digium developers when you have problems in Asterisk. There are developers that are not employees of Digium... There used to be more outside developers, but Digium is a great place to work... ;-) Hmm. Haven't found any Digium Stockholm office to discuss with ;-) /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. Any MySQL developers out there that can help us fix this? We need someone that has been developing towards the Mysql api. And please do not always refer to Digium developers when you have problems in Asterisk. There are developers that are not employees of Digium... There used to be more outside developers, but Digium is a great place to work... ;-) Hmm. Haven't found any Digium Stockholm office to discuss with ;-) That hasn't stopped any of the Canadian employees. :-) -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On May 5, 2008 01:58:42 pm Tilghman Lesher wrote: Hmm. Haven't found any Digium Stockholm office to discuss with ;-) That hasn't stopped any of the Canadian employees. :-) That's because nothing stops Canadians, short of Hockey Night in Canada :-) -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all. zaptel.conf Description: Binary data extensions.conf Description: Binary data sip.conf Description: Binary data zapata.conf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all. asterisk.rar Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help please
hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all.fxsks=1 loadzone=es defaultzone=es[general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [miprimerejemplo] exten = 2,1,Dial(SIP/2,30,Ttm) exten = 2,2,Hangup exten = 2,102,Voicemail(2) exten = 2,103,Hangup exten = 20100,1,Dial(SIP/20100,30,Ttm) exten = 20100,2,Hangup exten = 20100,102,Voicemail(20100) exten = 20100,103,Hangup exten = 20200,1,Dial(SIP/20200,30,Ttm) exten = 20200,2,Hangup exten = 202000,102,Voicemail(20200) exten = 20200,103,Hangup exten = 20300,1,Dial(SIP/20300,30,Ttm) exten = 20300,2,Hangup exten = 203000,102,Voicemail(20300) exten = 20300,103,Hangup exten = 20400,1,Dial(SIP/20400,30,Ttm) exten = 20400,2,Hangup exten = 204000,102,Voicemail(20400) exten = 20400,103,Hangup exten = 3,1,VoicemailMain exten = _9,1,Dial(SIP/[EMAIL PROTECTED]) exten = _9,2,Hangup[general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [2] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20100] type=friend secret=some qualify=yes nat=yes host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20200] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20300] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [20400] type=friend secret=some qualify=yes nat=no host=dynamic canreinvite=no context=miprimerejemplo [EMAIL PROTECTED] [VoipBuster] type=peer host=sip.voipbuster.com username=somesi3 fromuser=somesi3 secret=some[channels] language=es context=incoming switchtype=euroisdn usercallid=yes hidecallerid=no musiconhold=default callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes inmediate=no canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbriged=yes rxgain=0.0 txgain=0.0 group=1 signalling=fxs_ks context=incoming channel=4___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help please
On Tue, Apr 24, 2007 at 10:21:53AM +0200, Josu Lazkano Lete wrote: hello, I have a A400P01 PCI from OpenVox. I have installed some extension and a VoipBuste account to callo out of my LAN. How can I receive and send calls from a nd to outside by my analog line??? I want to receive dthe calls from 20100 extension. Here you have my config files, thanks for all. A few things unrelated to your issue that may help you to get more effetive answers from this list: 1. Please give more descriptive subject lines. The subject of your first message (asterisk on Debian) was good. The subject of your more recent messages are rather poor: please help me gives no hint as to what the problem is. 2. You have already started a thread, and another list member has asked you for some details. The files attached to this message appear to be replies to that message. If they are, please follow-up the same thread. 3. You did not write what is actually wrong: I do XYZ. I expect it to cause ABC but instead I get DEF See also the document on how to ask questions effectively: http://www.catb.org/~esr/faqs/smart-questions.html -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
-- Forwarded message --From:William Piper [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Sat, 30 Sep 2006 22:08:23 -0400 Subject:Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LANSure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try see what happens. bp On 9/28/06, Wolfgang_Borgon [EMAIL PROTECTED] wrote: David,Yes, I've also forwarded port 4569 to the server.Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as theclient isn't going out past the LAN, it shouldn'tmatter... unless there's something else going on thatI don't know about.ThanksWolfgang--- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote:- I've already sunk several hours into this without any real progress, so I'd really appreciate any helpMy task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP.I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host= 192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client)== You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? DaveIn your iax.conf, instead of...host= 192.168.1.40Use...host=dynamic-Buki - Da Naija Man ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
On 28 Sep 2006, at 21:39, Wolfgang_Borgon wrote: David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- You'll need to run ethereal or another packet sniffer on the network to see what is happening. In general if your asterisk is going to _register_ (or qualify) your external IAX provider you don't need to port-forward 4569, the registration will setup a suitable nat/port mapping and the qualify (or re-register) will keep it alive. (But that isn't your problem) T. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
Sure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try see what happens. bp On 9/28/06, Wolfgang_Borgon [EMAIL PROTECTED] wrote: David,Yes, I've also forwarded port 4569 to the server.Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as theclient isn't going out past the LAN, it shouldn'tmatter... unless there's something else going on thatI don't know about.ThanksWolfgang--- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote:- I've already sunk several hours into this without any real progress, so I'd really appreciate any helpMy task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP.I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client)== You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __Yahoo! Music UnlimitedAccess over 1 million songs. Try it free.http://music.yahoo.com/unlimited/ ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-use ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help please == Wrong password
Hi i have a small problems with my asterisk connected to phonesystems : Now i have this message: -- SIP read from 62.39.136.151:5060: SIP/2.0 403 Cant accept register from myself Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060 From: sip:[EMAIL PROTECTED];tag=as42b95c05 To: sip:[EMAIL PROTECTED];tag=e3fe971527b049ab0c1e91db33fcbf5f.cf8c Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Server: PSN Sip Proxy (1.1.3 (PRX3-EXTERNAL)) Content-Length: 0 Warning: 392 62.39.136.151:5060 Noisy feedback tells: pid=11434 req_src_ip=62.39.136.151 req_src_port=5060 in_uri=sip:sip3.phonesystems.net out_uri=sip:sip3.phonesystems.net via_cnt==2 --- (9 headers 0 lines)--- Aug 30 17:12:50 WARNING[15568]: chan_sip.c:10010 handle_response: Forbidden - wrong password on authentication for REGISTER but my login/password are correct into sip.conf the configuration have changed in asterisk 1.2.11 ? thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP PLEASE: What A Pain RSA
Hello all, I am having a terrible time with RSA registry. I have genetrated a key for 'pbx1too' with astgenkey and put it in /var/lib/asterisk/keys/, but when I reload iax2 I get an error: authenticate: Unable to find private key 'pbx1too' Where else would asterisk be looking for the key? I also put 'pbx1too.pub' in /var/lib/asterisk/keys/ I did give the key a passphrase, could this be the problem? Do I have to put the passphrase in some config file? Thanks, Murrah Boswell ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] HELP PLEASE: What A Pain RSA
OTR Comm wrote: I did give the key a passphrase, could this be the problem? Do I have to put the passphrase in some config file? If your keys have passphrases, then you must run the 'init keys' CLI command to enter the passphrase before the keys can be loaded. It's much simpler to just not use passphrases and instead protect your private key every other way that you can :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] HELP PLEASE: What A Pain RSA
Hello Kevin, If your keys have passphrases, then you must run the 'init keys' CLI command to enter the passphrase before the keys can be loaded. It's much simpler to just not use passphrases and instead protect your private key every other way that you can :-) Yeah I rebuilt my keys without a passphrase, but I still can't get the two boxes to register. On one box called pbx1 (xxx.xxx.xxx.2), I have in iax.conf: [general] ... register = pbx1:[EMAIL PROTECTED] ... [pbx1too] type=friend host=dynamic auth=rsa inkeys=pbx1too outkeys=pbx1 username=pbx1too context=outgoing Where the keys for pbx1 are pbx1.pub and pbx1.key, and a copy of pbx1too.pub is in /var/lib/asterisk/keys/ And then on another box called pbx1too (xxx.xxx.xxx.3), in the iax.conf: [general] ... register = pbx1too:[EMAIL PROTECTED] ... [pbx1] type=friend host=dynamic auth=rsa inkeys=pbx1 outkeys=pbx1too username=pbx1 context=incoming Where the keys for pbx1too are pbx1too.pub and pbx1too.key, and a copy of pbx1.pub is in /var/lib/asterisk/keys/ I can't see anything wrong with this, but I have been at it for more hours than I want to think about right now :) Hope you can! Got another problem involving D-channels that go away and calls get dropped, then the D-channels come back and the B-channels get rebuilt. Only on PRI - SIP calls, but, one problem at a time. Thanks for looking, Murrah Boswell ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Help please
Hello, i`m carlos, i`m just begining to use Asterisk at Home, so i have learned to configure a several extensions, but now i have a FXO target and i wanna to connect to PSTN, but i dunno how to do. i`d like to receive support from you...thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) Here's the output of 'iax show peers': Name/UsernameHost Mask Port Status voxee/# 66.246.246.52 (S) 255.255.255.255 4569 Unmonitored client/client 192.168.1.40 (S) 255.255.255.255 0 Unmonitored demo/asterisk216.207.245.47 (S) 255.255.255.255 4569 Unmonitored Note the port listed at 0. Debug reponse to 'dial [EMAIL PROTECTED]': -- Executing Dial(ALSA/default, IAX2/client) in new stack -- Called client Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 1 DCall: 0 [192.168.1.40:0] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ilbc|ulaw|alaw|gsm) CALLING PRESNTN : 67 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en USERNAME: client FORMAT : 2 CAPABILITY : 64526 ADSICPE : 0 DATE TIME : 2005-10-10 00:04:14 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 1 DCall: 0 [192.168.1.40:0] VERSION : 2 CALLED NUMBER : s CODEC_PREFS : (ilbc|ulaw|alaw|gsm) CALLING PRESNTN : 67 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en USERNAME: client FORMAT : 2 CAPABILITY : 64526 ADSICPE : 0 DATE TIME : 2005-10-10 00:04:14 -- IAX2/client-1 is circuit-busy Oct 10 00:04:19 NOTICE[3615]: chan_iax2.c:2754 auto_congest: Auto-congesting cal l due to slow response -- Hungup 'IAX2/client-1' == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'ALSA/default' status is 'CONGESTION' __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David: Also port 1:2 is a good idea to forward to the server as well.. David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David: Also port 1:2 is a good idea to forward to the server as well.. Only needed for SIP. 4569 is all that is required for IAX2. David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang You might try: - [2278] type=friend secret=** host=dynamic context=clientcon --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] help please
what i have at present is asterisk installed on fedora core 3, with a dev kit with 1 fxo and 1fxs module and a TE110p card, using the sample files i can dial asterisk using the dev kit and get the asterisk welcome and congratulations message i have got the zaptel.conf conigured as follows fxoks=1fxsks=4 span=1,1,0,ccs,hdb3,crc4bchan=5-19,21-35dchan=20 loadzone=nldefaultzone=nl the zapata.conf as follows [channels]language=nlcontext=defaultusecallerid=yeshidecallerid=nocallwaiting=yescallwaitinguserid=yesthreewaycalling=yesechocancel=yesechocancelwhenbridged=norxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nocallerid=206388230busydetect=nocallprogress=nomusiconhold=default signalling=fxo_kschannel = 1 signalling fxs_kschannel = 4 what i want to be able to do is this have 5incommingnumbers over thepri(e1) lines and also dialing out over whichever line is available on the pri(e1) line. i would like to know 1/ how do you setup your zapata.conf file for 2 cards??? as i have 1 dev kit and 1 te110p card 2/ how do i get asterisk to detect what number has ben dialled by the outside caller and route it to the appropriate extension, if no answer after 10 seconds ring all the other phones and if no answer after that go to voicemail? and allow other users to pick up someone elses phone from their extension. user10031204161092 (full international number)0204161092 (number as dialled from the netherlands)4161092 (number as dialled from amsterdam) user20031204161091 (full international number)0204161091 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) user30031204161093 (full international number)0204161093 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) user40031204161094 (full international number)0204161094 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) 3/ how do i record calls and set them to a file in the format extensionnmrdatetime.mp3 I know its a lot to ask but once i get it up and running i intend on providing a step by step method of how i installed it from installing fdc3 to up and running so will help many others too so please help. ___ 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
[Asterisk-Users] Help Please Multiple Users for Broadvoice
I would like to be able to have multiple users (the wife and kids) to be able to access the Broadvoice account at the same. No complaining that way from them J. I seen someones configuration in the group here but now I cant find it (lost my glasses). If someone could post theirss or the shortcut that would be great. Thanks for your help. Dad shes on the phone again ! ___ 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
RE: [Asterisk-Users] Help Please Multiple Users for Broadvoice
http://geekgazette.com has a "how to" for Broadvoice -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mr. barkerSent: Saturday, May 14, 2005 6:03 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Help Please Multiple Users for Broadvoice I would like to be able to have multiple users (the wife and kids) to be able to access the Broadvoice account at the same. No complaining that way from them J. I seen someones configuration in the group here but now I cant find it (lost my glasses). If someone could post theirss or the shortcut that would be great. Thanks for your help. Dad shes on the phone again ! ___ 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
[Asterisk-Users] Help please for newb on Asterisk to Vonage
I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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
RE: [Asterisk-Users] Help please for newb on Asterisk to Vonage
You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage IOf you would read his original posting, you will see he wants to connect with Asterisk via his softphone account, and make outgoing calls through his ATA and X100 card. All should be doable. Someone posted a working config to the list within the past 10 days or so, so a search through the recent archives should bring a working result. The key here is the MAIN Vonage number is locked to the ATA, but a softphone add on account from Vonage should work. John Novack If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I was only able to get the softphone account to make inbound calls on one sip.conf config, or outbound calls on another sip.conf config. I didnt investigate the issue completely, but from what I could tell, they wouldnt allow multiple SIP sessions from the same IP address. I didnt try running 2 asterisk instances (one for in and one for out) that would talk to each other, however I would believe that Vonage does account state/session checks. Net2Phone now does authentication on 3 credentials; account number, pin, and MAC address. Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
Im a newbie to this list (joined today). Other than Broadvoice, what voip providers work well with Asterisk? I'd like a service that will allow trunking so that I can have more than one outbound/inbound call if possible. JD Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Help please for newb on Asterisk to Vonage
Good point, I forgot about using the softphone add-on account. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Tuesday, March 22, 2005 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage IOf you would read his original posting, you will see he wants to connect with Asterisk via his softphone account, and make outgoing calls through his ATA and X100 card. All should be doable. Someone posted a working config to the list within the past 10 days or so, so a search through the recent archives should bring a working result. The key here is the MAIN Vonage number is locked to the ATA, but a softphone add on account from Vonage should work. John Novack If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
Check the list archive. This thread just happened a couple days ago. A couple that I remember as supporting IAX: voipjet.com nufone.net You also might try opbx.com JD Austin wrote: Im a newbie to this list (joined today). Other than Broadvoice, what voip providers work well with Asterisk? I'd like a service that will allow trunking so that I can have more than one outbound/inbound call if possible. JD Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
I made the Vonage mistake too. Cost me a hundred bucks! They will hit you with a Disconnect Fee, if you don't return their equipment in the original box. Sux, if you ask me. Vonage does offer an attractive flat rate for 500 minutes. You could, purchase a digium card, and plug the Vonage POTS line into the digium card, just like you would a POTS line. Kinda a workaround, but I tried it and it works. Or you can buy the softfone addon, and pay more per month to Vonage and settle for SIP... Or better yet, try: http://connect.voicepulse.com You get free incoming calls. A low monthly rate, and a cheap per minute rate for outgoing. Best part of all, is you get trunking.. 4 simultaneous calls in and out. Which is what I think you said you were looking for. Check the list archive. This thread just happened a couple days ago. A couple that I remember as supporting IAX: voipjet.com nufone.net You also might try opbx.com JD Austin wrote: Im a newbie to this list (joined today). Other than Broadvoice, what voip providers work well with Asterisk? I'd like a service that will allow trunking so that I can have more than one outbound/inbound call if possible. JD Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Help Please!!!!
Thanks, I will begin my testing Erick - Original Message - From: Race Vanderdecken [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 8:18 PM Subject: RE: [Asterisk-Users] Help Please Greetings Mr. Weber, Remember the rule in mathematics that is much easier to solve for one variable. You stateed you are having a problem with the 1088 extension. If look like you are trying to make a call from the 404 extension to the 1088 extension. 1. If you have 6 ATA's running shut 5 of them off. Test each one separately. Then turn one on at a time and see the problem can be traced to one ATA 2. You are getting sent an authorization request from asterisk to the 1088 extension. WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Make sure you don't have any of the secret= or the md5secret= stuff set in the sip.conf, until you can get each phone to talk in the open. Then change, one, 1, uno, phone at a time. 3. If you have a SIP phone that is not an ATA then set it up and try to dial the 1088 and see if you get the same thing. 4. Do a sip show users to make sure the 1088 is registered with asterisk. 5. Do the normal, things don't work dance, by unplugging the phone and reconnecting a different phone to the ata. Change the power suplly with another ata. Change the RJ45 patch cable. Try a different port in the switch or wall. Swap one of the known working ATA and change it to the 1088 ata. 6. Go to lunch and have a beer. Find a new job and settle down with a good woman. Leave telecom and go into organic farming. Race The Tyrant Vanderdecken [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Wednesday, February 16, 2005 2:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help Please Importance: High I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1
[Asterisk-Users] Help Please!!!!
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED];expires=120 Date: Wed, 16 Feb 2005 00:43:46 GMT Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7 From: asterisk sip:[EMAIL PROTECTED];tag=as59adf4c2 To: sip:201.133.170.82 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 201.133.170.82:5060 Destroying call '[EMAIL PROTECTED]' set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060
RE: [Asterisk-Users] Help Please!!!!
Greetings Mr. Weber, Remember the rule in mathematics that is much easier to solve for one variable. You stateed you are having a problem with the 1088 extension. If look like you are trying to make a call from the 404 extension to the 1088 extension. 1. If you have 6 ATA's running shut 5 of them off. Test each one separately. Then turn one on at a time and see the problem can be traced to one ATA 2. You are getting sent an authorization request from asterisk to the 1088 extension. WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Make sure you don't have any of the secret= or the md5secret= stuff set in the sip.conf, until you can get each phone to talk in the open. Then change, one, 1, uno, phone at a time. 3. If you have a SIP phone that is not an ATA then set it up and try to dial the 1088 and see if you get the same thing. 4. Do a sip show users to make sure the 1088 is registered with asterisk. 5. Do the normal, things don't work dance, by unplugging the phone and reconnecting a different phone to the ata. Change the power suplly with another ata. Change the RJ45 patch cable. Try a different port in the switch or wall. Swap one of the known working ATA and change it to the 1088 ata. 6. Go to lunch and have a beer. Find a new job and settle down with a good woman. Leave telecom and go into organic farming. Race The Tyrant Vanderdecken [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Wednesday, February 16, 2005 2:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help Please Importance: High I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL
[Asterisk-Users] Help please
Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
in my opinion, asterisk is the best solution. it is relatively very cheap, asterisk is free and u can use any soft phones to do calls. you can find cheap digium cards, around $400+. you can set up any normal phone to FXS cards. yu can do conference call and voicemails in asterisk. this link may be helpful to read though there are lots of it out there http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb --- Michael Welter [EMAIL PROTECTED] wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
in my opinion, asterisk is the best solution. it is relatively very cheap, asterisk is free and u can use any soft phones to do calls. you can find cheap digium cards, around $400+. you can set up any normal phone to FXS cards. yu can do conference call and voicemails in asterisk. this link may be helpful to read though there are lots of it out there http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb --- Michael Welter [EMAIL PROTECTED] wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
On Wed, 2003-11-19 at 10:52, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
On Wed, 2003-11-19 at 10:52, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Sorry for the last empty message, don't know if I was able to delete it from my outbox quick enough. Anyways, we discussed pricing recently. Your price for lines is pretty nice. You probably couldn't justify the extra cost of switching to T1 or PRI as the loop itself would cost a little more than you currently are paying. So now you have to figure out how to deal with these lines. You will have to decide whether or not you want IP phones, or analog phones. Either way, you probably are looking at a channel bank and a T1 card to get the 8 lines into a single asterisk machine. 8x12 would fit on a Zhone, but you have to know that the zhone doesn't pass callerid on the FXO ports. You could go with just about any modular channel bank that supports FXO ports. Look at the Adtrans or Adit lines. So for an analog system you are looking at $500 PC $500 T100P card $500 or so for a channel bank of of ebay. $30x12 for phones. or approximately $1860 if you get lucky on ebay. On a purely money view point, if you don't for see growth in the number of lines into your office, you won't see the break even of asterisk for about 4 years over the Centrex lines. Of course you can assign value to having your own machine in house, and the VoIP possibilities, and you can claim break even in a shorter term. Don't take this as downing asterisk, I love it. I just wanted to cut to the dollars since you have placed them on the table. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
On Wed, Nov 19, 2003 at 09:52:13AM -0700, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are I have a client running 9 incomming PRI lines with their 768K internet all on one T-1. Works very well. I have another client with 4 pots lines. Works OK, but you have some problems with call detection. voice mailboxes on each line. Voice mail is excellent with Asterisk. Very flexible. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. With Asterisk you would be looking at an initial fixed price investment and about 20% of that per year for support. The cost for pots lines vs. T1 for the PBX would be about the same. You would need to check with LECs and CLECs to determine the most economical way to deliver the PSTN services to your office. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Absolutely consider Asterisk. You can use standard POTs style phones, some ADSI phones and SIP phones. The Cisco 7960 is very nice. You can set up Meetme conferences and you can conference many incoming calls to the same connection. If you have VoIP channels you could conference them in as well. The client I mentioned above using the T1 PRI does depositions on the conferences. We've had conferences up for 7 hours! Yes, Voicemail is very good and flexible. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help please with single t1 configuration
-Original Message- From: Barry Porch [mailto:[EMAIL PROTECTED] Yes, I run ztcfg -vvv and it tells me that there are 0 channels. Barry, are you modifying the zaptel.conf file in /etc or /etc/asterisk. If the latter, try ztcfg -vvv -c /etc/asterisk/zaptel.conf. I had a similar issue last week when first starting with Asterisk. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help please with single t1 configuration
Yes, I run ztcfg -vvv and it tells me that there are 0 channels. -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: Saturday, August 09, 2003 6:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] help please with single t1 configuration did you happen to run ztcfg after you setup the configs? On Sat, 9 Aug 2003, Barry Porch wrote: I am attempting to set up an Asterisk box which I am only concerned with getting a single T1 working. I have this T1 connected to my PBX and I am looking at using Asterisk as a conference bridge. Here is my zaptel.conf: span=1,1,0,d4,ami em=1-24 loadzone=us defaultzone=us Here is my Zapata.conf: [channels] language=en context=default signalling=em_w group=1 channel=1-24 As you can see I am trying to set this up as a D4/AMI EM T1. I also have modprobe zaptel and modprobe wct1xxp in my rc.local. When I try to run Asterisk it complains that it cannot access channel 1 and dies. I can run zttool which sees the T100 card that I have installed. However, when I run ztcfg - it sells me that 0 channels are configured. What am I missing here? Thanks! Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help please with single t1 configuration
did you happen to run ztcfg after you setup the configs? On Sat, 9 Aug 2003, Barry Porch wrote: I am attempting to set up an Asterisk box which I am only concerned with getting a single T1 working. I have this T1 connected to my PBX and I am looking at using Asterisk as a conference bridge. Here is my zaptel.conf: span=1,1,0,d4,ami em=1-24 loadzone=us defaultzone=us Here is my Zapata.conf: [channels] language=en context=default signalling=em_w group=1 channel=1-24 As you can see I am trying to set this up as a D4/AMI EM T1. I also have modprobe zaptel and modprobe wct1xxp in my rc.local. When I try to run Asterisk it complains that it cannot access channel 1 and dies. I can run zttool which sees the T100 card that I have installed. However, when I run ztcfg - it sells me that 0 channels are configured. What am I missing here? Thanks! Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help please with single t1 configuration
I am attempting to set up an Asterisk box which I am only concerned with getting a single T1 working. I have this T1 connected to my PBX and I am looking at using Asterisk as a conference bridge. Here is my zaptel.conf: span=1,1,0,d4,ami em=1-24 loadzone=us defaultzone=us Here is my Zapata.conf: [channels] language=en context=default signalling=em_w group=1 channel=1-24 As you can see I am trying to set this up as a D4/AMI EM T1. I also have modprobe zaptel and modprobe wct1xxp in my rc.local. When I try to run Asterisk it complains that it cannot access channel 1 and dies. I can run zttool which sees the T100 card that I have installed. However, when I run ztcfg it sells me that 0 channels are configured. What am I missing here? Thanks! Barry
