Re: [Freeswitch-users] dialplan_hunt taking too long
Thanks for the tip with f8. I did a call from soft phone (x-lite) 1003 to extension 5000. This is the log: http://pastebin.freeswitch.org/5791 So it seems, that dialplan_hunt is ok. As you can see, there is a gap ~8 seconds between line 127 and 128. The PC is pentium 3, 1.8 GHz, which is pretty idle. It is running Debian etch x86 on kernel 2.6.18-6-686. I also made the same test on windows xp, and on other notebook running Debian Etch 64bit, and the gap was smaller ( 1-2 seconds). I also did run freeswitch using gdb, and after I dialled extension 5000, I waited for a few seconds, and then I hit ctrl+c and made backtraces from all threads: http://pastebin.freeswitch.org/5792 sample Wireshark trace on client side when dialling out extension 5000 (client is 163.242.69.92, server is 172.16.3.40): http://pastebin.freeswitch.org/5793 My guess would be the \'while\' inside switch_rtp_set_local_address(), but I prefer to hear the expert opinion :-). Thank you. - Original message- Od: \Anthony Minessale\ Komu: Poslaná: 10.10.2008 04:05 Predmet: Re: [Freeswitch-users] dialplan_hunt taking too long what extension is it hitting? The entire log with debug on would be more useful (press f8 on the console and try the call again) __ Sprievodca herným svetom - http://hry.sme.sk/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] gateway with TLS
Hi, I can't access IRC, because its forbiden. I have freeswitch like endpoint (gateway); I have an IPphone with TLS; When i make a call using the IPphone with TLS to my freeswitch (9995; delay_echo), works all signaling with TLS; When i make a call using freeswitch to my ipphone, when i answer the call, HANGUP! Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan_hunt taking too long
From the looks of your SDP on line 129 your system appears to be misconfigured with IPv6 Try disabling IPv6 on your interface or setting sofia up according to the IPv6 instructions on the wiki. 2008/10/10 Ineya Ineya [EMAIL PROTECTED] Thanks for the tip with f8. I did a call from soft phone (x-lite) 1003 to extension 5000. This is the log: http://pastebin.freeswitch.org/5791 So it seems, that dialplan_hunt is ok. As you can see, there is a gap ~8 seconds between line 127 and 128. The PC is pentium 3, 1.8 GHz, which is pretty idle. It is running Debian etch x86 on kernel 2.6.18-6-686. I also made the same test on windows xp, and on other notebook running Debian Etch 64bit, and the gap was smaller ( 1-2 seconds). I also did run freeswitch using gdb, and after I dialled extension 5000, I waited for a few seconds, and then I hit ctrl+c and made backtraces from all threads: http://pastebin.freeswitch.org/5792 sample Wireshark trace on client side when dialling out extension 5000 (client is 163.242.69.92, server is 172.16.3.40): http://pastebin.freeswitch.org/5793 My guess would be the \'while\' inside switch_rtp_set_local_address(), but I prefer to hear the expert opinion :-). Thank you. - Original message- Od: \Anthony Minessale\ Komu: Poslaná: 10.10.2008 04:05 Predmet: Re: [Freeswitch-users] dialplan_hunt taking too long what extension is it hitting? The entire log with debug on would be more useful (press f8 on the console and try the call again) __ Sprievodca herným svetom - http://hry.sme.sk/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ODBC through JS
Guys please help me on this
I am still getting that error
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified*
I can connect to the database successfully using isql -v command
I can't figure out whats going on
Here's my script:
use(ODBC);
var DSN=PostgreSQL;
var DB_USER=username;
var DB_PASS=password;
var db = new ODBC(DSN, DB_USER, DB_PASS);
//console_log(notice,DB.path+\n);
if(db.connect()){
console_log(notice,Connected to DB);
}
else{
console_log(notice,Still probs);
}
*Here's the output:*
[EMAIL PROTECTED] 2008-10-10 15:41:32 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
2008-10-10 15:41:32 [NOTICE] testdb.js:1 console_log() Still probs
Please Help me
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 5:37 PM, Gayatri Kulkarni [EMAIL PROTECTED]wrote:
Guys,
I am now having this problem:
when i run my script i get this error -
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
my .*./etc/odbc.ini* looks like this:
[postgres]
Driver = PostgreSQL
SERVER = localhost
PORT = 5432
DATABASE = postgres
UserName = username
Password = password
and .*./etc/odbcinst.ini:*
[ODBC]
Trace = yes
TraceFile = /tmp/trace.log
[PostgreSQL]
Description = PostgreSQL driver
Driver = /usr/local/lib/libodbpsql.so
Setup = /usr/local/lib/libodbpsqlS.so
UsageCount = 100
--
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 11:32 AM, Brian West [EMAIL PROTECTED] wrote:
thats because you don't load it in modules.conf.xml you load it
in conf/autoload_configs/spidermonkey.conf.xml
/b
On Oct 7, 2008, at 12:52 AM, Gayatri Kulkarni wrote:
Still the same :(
It builds installs the shard library at {prefix}/freeswitch/lib directory
while all the modules .so files are in {prefix}/freeswitch/mod directory
If i copy the shared library to the mod directory, i get an error saying
Invalid
ELF Header !!
--
Regards,
Gayatri Kulkarni
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Regards,
Gayatri Kulkarni
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Re: [Freeswitch-users] recording in telnet
Hi, We can use the recording in api as the one like the below, api uuid_record uuid start/path to record the file. Thanks On Thu, Oct 9, 2008 at 5:37 PM, Michael Jerris [EMAIL PROTECTED] wrote: On Oct 9, 2008, at 8:01 AM, Gopal krishnan wrote: Hi, I am trying to record thru telnet with sendevent record and also tried sendevent record_session but I cant able to record. Is there any command to record thru telnet? http://wiki.freeswitch.org/wiki/Event_Socket#SendMsg http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record Also not that you can not interact with the media like this if you are using proxy media or bypass media modes. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] gateway with TLS
When i make call without TLS (pa call sip:[EMAIL PROTECTED][EMAIL PROTECTED]== works) to my ipphone works!!! Just when i use freeswitch tls to call to my ipphone don't work (pa call sips:[EMAIL PROTECTED][EMAIL PROTECTED]) !!! thanks On Fri, Oct 10, 2008 at 10:02 AM, paulo leonardo [EMAIL PROTECTED] wrote: Hi, I can't access IRC, because its forbiden. I have freeswitch like endpoint (gateway); I have an IPphone with TLS; When i make a call using the IPphone with TLS to my freeswitch (9995; delay_echo), works all signaling with TLS; When i make a call using freeswitch to my ipphone, when i answer the call, HANGUP! Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] General voice mail boxes and other questions
Is there anyone who could give me an answer to these questions, or is my questions too generalized? I am trying to completa a general IVR and I see Freeswitch definitely on the next gen path for this industry. Thanks Damon Brown wrote: Hello all, As a newbee but expert in other nameless PBX's, I was wondering the following: 1. What approach in the xml config files you would use to create general mailboxes that multiple extensions have access too and need indicators for? 2. I was using the directory.lua example located in the wiki to create a directory and it works up to the pount of matching the extension. Once the caller presses confirm for the name and it returns to no directories match your search. I know this should be to the author of that example, but I was hoping someone may have experience with it. I do have mod_lua it runs through most of the logic, the example I used was: http://wiki.freeswitch.org/wiki/Examples_directory_lua 3. I would like to write a daemon that interfaces with freeswitch outside of the XML matching. For example, one that will connect to a DB, pick up a caller and send conference participants in based on their status. Is this possible or are do all scripts only run within the dialplan? Thanks for your time in answering these in advance. Best regards, Damon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org begin:vcard fn:Damon Brown n:Brown;Damon org:Technicate Solutions;Development and Solutions adr:Suite K;;7529 Sunset Avenue;Fair Oaks;CA;95628;USA email;internet:[EMAIL PROTECTED] title:Owner tel;work:1.916.880.1128 tel;cell:1.916.601.8190 url:www.technicate.com version:2.1 end:vcard ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] gateway with TLS
do you have your dialplan setup to parse sips extensions? On Fri, Oct 10, 2008 at 8:34 AM, paulo leonardo [EMAIL PROTECTED] wrote: When i make call without TLS (pa call sip:[EMAIL PROTECTED][EMAIL PROTECTED]== works) to my ipphone works!!! Just when i use freeswitch tls to call to my ipphone don't work (pa call sips:[EMAIL PROTECTED][EMAIL PROTECTED]) !!! thanks On Fri, Oct 10, 2008 at 10:02 AM, paulo leonardo [EMAIL PROTECTED] wrote: Hi, I can't access IRC, because its forbiden. I have freeswitch like endpoint (gateway); I have an IPphone with TLS; When i make a call using the IPphone with TLS to my freeswitch (9995; delay_echo), works all signaling with TLS; When i make a call using freeswitch to my ipphone, when i answer the call, HANGUP! Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] gateway with TLS
how can i do that? is there documentation in wiki.freeswitch.org about this? thanks On Fri, Oct 10, 2008 at 11:12 AM, Anthony Minessale [EMAIL PROTECTED] wrote: do you have your dialplan setup to parse sips extensions? On Fri, Oct 10, 2008 at 8:34 AM, paulo leonardo [EMAIL PROTECTED] wrote: When i make call without TLS (pa call sip:[EMAIL PROTECTED][EMAIL PROTECTED]== works) to my ipphone works!!! Just when i use freeswitch tls to call to my ipphone don't work (pa call sips:[EMAIL PROTECTED][EMAIL PROTECTED]) !!! thanks On Fri, Oct 10, 2008 at 10:02 AM, paulo leonardo [EMAIL PROTECTED]wrote: Hi, I can't access IRC, because its forbiden. I have freeswitch like endpoint (gateway); I have an IPphone with TLS; When i make a call using the IPphone with TLS to my freeswitch (9995; delay_echo), works all signaling with TLS; When i make a call using freeswitch to my ipphone, when i answer the call, HANGUP! Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ODBC through JS
either value for DSN name gives the same result :(
On Fri, Oct 10, 2008 at 4:27 PM, Jonas Gauffin [EMAIL PROTECTED]wrote:
your DSN name is postgres and not PostgreSQL
On Fri, Oct 10, 2008 at 12:34 PM, Gayatri Kulkarni [EMAIL PROTECTED]
wrote:
Guys please help me on this
I am still getting that error
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified*
I can connect to the database successfully using isql -v command
I can't figure out whats going on
Here's my script:
use(ODBC);
var DSN=PostgreSQL;
var DB_USER=username;
var DB_PASS=password;
var db = new ODBC(DSN, DB_USER, DB_PASS);
//console_log(notice,DB.path+\n);
if(db.connect()){
console_log(notice,Connected to DB);
}
else{
console_log(notice,Still probs);
}
*Here's the output:*
[EMAIL PROTECTED] 2008-10-10 15:41:32 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
2008-10-10 15:41:32 [NOTICE] testdb.js:1 console_log() Still probs
Please Help me
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 5:37 PM, Gayatri Kulkarni [EMAIL PROTECTED]wrote:
Guys,
I am now having this problem:
when i run my script i get this error -
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
my .*./etc/odbc.ini* looks like this:
[postgres]
Driver = PostgreSQL
SERVER = localhost
PORT = 5432
DATABASE = postgres
UserName = username
Password = password
and .*./etc/odbcinst.ini:*
[ODBC]
Trace = yes
TraceFile = /tmp/trace.log
[PostgreSQL]
Description = PostgreSQL driver
Driver = /usr/local/lib/libodbpsql.so
Setup = /usr/local/lib/libodbpsqlS.so
UsageCount = 100
--
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 11:32 AM, Brian West [EMAIL PROTECTED]wrote:
thats because you don't load it in modules.conf.xml you load it
in conf/autoload_configs/spidermonkey.conf.xml
/b
On Oct 7, 2008, at 12:52 AM, Gayatri Kulkarni wrote:
Still the same :(
It builds installs the shard library at {prefix}/freeswitch/lib
directory
while all the modules .so files are in {prefix}/freeswitch/mod directory
If i copy the shared library to the mod directory, i get an error saying
Invalid ELF Header !!
--
Regards,
Gayatri Kulkarni
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--
Regards,
Gayatri Kulkarni
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Gayatri Kulkarni
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Re: [Freeswitch-users] ODBC through JS
your DSN name is postgres and not PostgreSQL
On Fri, Oct 10, 2008 at 12:34 PM, Gayatri Kulkarni
[EMAIL PROTECTED]wrote:
Guys please help me on this
I am still getting that error
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified*
I can connect to the database successfully using isql -v command
I can't figure out whats going on
Here's my script:
use(ODBC);
var DSN=PostgreSQL;
var DB_USER=username;
var DB_PASS=password;
var db = new ODBC(DSN, DB_USER, DB_PASS);
//console_log(notice,DB.path+\n);
if(db.connect()){
console_log(notice,Connected to DB);
}
else{
console_log(notice,Still probs);
}
*Here's the output:*
[EMAIL PROTECTED] 2008-10-10 15:41:32 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
2008-10-10 15:41:32 [NOTICE] testdb.js:1 console_log() Still probs
Please Help me
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 5:37 PM, Gayatri Kulkarni [EMAIL PROTECTED]wrote:
Guys,
I am now having this problem:
when i run my script i get this error -
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
my .*./etc/odbc.ini* looks like this:
[postgres]
Driver = PostgreSQL
SERVER = localhost
PORT = 5432
DATABASE = postgres
UserName = username
Password = password
and .*./etc/odbcinst.ini:*
[ODBC]
Trace = yes
TraceFile = /tmp/trace.log
[PostgreSQL]
Description = PostgreSQL driver
Driver = /usr/local/lib/libodbpsql.so
Setup = /usr/local/lib/libodbpsqlS.so
UsageCount = 100
--
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 11:32 AM, Brian West [EMAIL PROTECTED] wrote:
thats because you don't load it in modules.conf.xml you load it
in conf/autoload_configs/spidermonkey.conf.xml
/b
On Oct 7, 2008, at 12:52 AM, Gayatri Kulkarni wrote:
Still the same :(
It builds installs the shard library at {prefix}/freeswitch/lib directory
while all the modules .so files are in {prefix}/freeswitch/mod directory
If i copy the shared library to the mod directory, i get an error saying
Invalid ELF Header !!
--
Regards,
Gayatri Kulkarni
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--
Regards,
Gayatri Kulkarni
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[Freeswitch-users] Strange originate behavior in xml_rpc
Hi,
I've been testing some xml_rpc scripts to make calls. (For a click
to call application I want to write.)
I'm experiencing some strange behavior in regards to setting the
caller id.
If I DON't pass a caller id with the originate command, the calls
works perfectly and the caller id shows as 000-000-.
If I DO pass a caller id variable, then I still get a call ringing,
the caller id is still 000-000- and when I answer, it hangs up
immediately and the debug in freeswitch shows an error. I'm getting a
407 Proxy Authentication Required error.
I'm trying to understand why setting a caller_id triggers an error
when not setting one works. Could it be something setup wrong in my
dialplan?
Here is the effective part of the XML I'm passing to Freeswitch:
valuestring{effective_caller_id_number=3235551212,accountcode=1}sofia/internal/[EMAIL
PROTECTED]
amp;bridge(sofia/internal/313555%111.111.111.111)/string/
value
Thanks,
-Noah
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Re: [Freeswitch-users] ODBC through JS
latest version of odbc.ini:
[ODBC Data Sources]
PostgreSQL = PostgreSQL Database Driver
[PostgreSQL]
Description = PostgreSQL Database Driver
Driver = /usr/local/lib/libodbcpsql.so
Trace = Yes
TraceFile = sql.log
Database= postgres
Servername = 58.68.117.43
UserName= username
Password= password
Port= 5432
Protocol= 6.4
ReadOnly= No
RowVersioning = No
ShowSystemTables= No
ShowOidColumn = No
FakeOidIndex= No
ConnSettings=
[Default]
Driver = /usr/local/lib/libodbcpsql.so
[ODBC]
Trace = 1
TraceFile = /var/log/odbctrace.out
Driver = /usr/local/lib/libodbcpsql.so
On Fri, Oct 10, 2008 at 5:28 PM, Gayatri Kulkarni [EMAIL PROTECTED]wrote:
either value for DSN name gives the same result :(
On Fri, Oct 10, 2008 at 4:27 PM, Jonas Gauffin [EMAIL PROTECTED]wrote:
your DSN name is postgres and not PostgreSQL
On Fri, Oct 10, 2008 at 12:34 PM, Gayatri Kulkarni
[EMAIL PROTECTED] wrote:
Guys please help me on this
I am still getting that error
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified*
I can connect to the database successfully using isql -v command
I can't figure out whats going on
Here's my script:
use(ODBC);
var DSN=PostgreSQL;
var DB_USER=username;
var DB_PASS=password;
var db = new ODBC(DSN, DB_USER, DB_PASS);
//console_log(notice,DB.path+\n);
if(db.connect()){
console_log(notice,Connected to DB);
}
else{
console_log(notice,Still probs);
}
*Here's the output:*
[EMAIL PROTECTED] 2008-10-10 15:41:32 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
2008-10-10 15:41:32 [NOTICE] testdb.js:1 console_log() Still probs
Please Help me
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 5:37 PM, Gayatri Kulkarni [EMAIL PROTECTED]
wrote:
Guys,
I am now having this problem:
when i run my script i get this error -
[EMAIL PROTECTED] 2008-10-07 17:27:24 [ERR] switch_odbc.c:160
switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified
my .*./etc/odbc.ini* looks like this:
[postgres]
Driver = PostgreSQL
SERVER = localhost
PORT = 5432
DATABASE = postgres
UserName = username
Password = password
and .*./etc/odbcinst.ini:*
[ODBC]
Trace = yes
TraceFile = /tmp/trace.log
[PostgreSQL]
Description = PostgreSQL driver
Driver = /usr/local/lib/libodbpsql.so
Setup = /usr/local/lib/libodbpsqlS.so
UsageCount = 100
--
Regards,
Gayatri Kulkarni
On Tue, Oct 7, 2008 at 11:32 AM, Brian West [EMAIL PROTECTED]wrote:
thats because you don't load it in modules.conf.xml you load it
in conf/autoload_configs/spidermonkey.conf.xml
/b
On Oct 7, 2008, at 12:52 AM, Gayatri Kulkarni wrote:
Still the same :(
It builds installs the shard library at {prefix}/freeswitch/lib
directory
while all the modules .so files are in {prefix}/freeswitch/mod
directory
If i copy the shared library to the mod directory, i get an error
saying Invalid ELF Header !!
--
Regards,
Gayatri Kulkarni
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Regards,
Gayatri Kulkarni
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--
Regards,
Gayatri Kulkarni
--
Regards,
Gayatri Kulkarni
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Re: [Freeswitch-users] VOIP vs PSTN
On Oct 10, 2008, at 10:54 AM, Alfred Richmond wrote:
Hello,
I am attempting to generate a message to convert to speech and send
it out to my users. I am a newbie but I am just not getting it after
reading through the documentation. In testing it works fine when
sending to my voip connected users using the following:
bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/
internal/1001 playback(/usr/local/freeswitch/sounds/warning.wav)
however, when I dial a cell phone as below it rings the phone but
immediately hangs up before playing the message. Is there something
obvious I am doing wrong?
bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/
gateway/sip.startec.com/1443111 playback(/usr/local/freeswitch/
sounds/warning.wav)
and then the follow up question is do I need bridge the call in the
dialplan like so?
!-- Dial 11 digit number via startec --
extension name=outbound
condition field=destination_number expression=^(\d{11})$
!--action application=set
data=effective_caller_id_number=4439951026/--
!-- action application=answer/
action application=playback data=/usr/local/freeswitch/
sounds/warning.wav/
--
!--action application=speak data=cepstral|david|Please
hold this is a test/ --
action application=bridge data=sofia/gateway/sip.startec.com/[EMAIL PROTECTED]
:5061/
/condition
/extension
With just this information, my guess is your call is not hitting the
same dialplan context. Turn the log output up to debug and see what
it is saying, my guess is its falling off the end of the public
context with no matching extension.
Mike
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[Freeswitch-users] SIP documentation, interop, etc
Hello Freeswitch users, I've started working on a wiki for SIP documentation for interop and other features. I've created a basic page for Freeswitch: http://www.submityoursip.com/wiki/Freeswitch Would someone with more Freeswitch SIP experience help me fill in some more details? Things like support for various specs, issues when working with other devices, etc. I'm still trying to work on a template for implementations and some sort of format but I figured I'd get some information up and see where it goes. Thanks, and let me know if you have any suggestions! P.S. - Obviously it's not just limited to Freeswitch... If you have experience with other SIP implementations, service providers, etc I'd love to read about it! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problem with first use.
Hello, I am new to freeswitch. Just installed freeswitch-1.0.1 with default configurations on CentOS 4.6 with OpenVZ. I registered two Cisco7960 phones using static IP with default users(1001.xml, 1007.xml). The registration are OK, but when I try to call from each other, it immedially send to voicemail. The following is the log info from console: (I replaced the real IP with 10.1.1.117 in the log) 2008-10-10 11:41:02 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes [bfe75310-638d-48df-8f3d-262143c15b22] 2008-10-10 11:41:02 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/[EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/[EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes) Ended 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes [CS_HANGUP] Is it possible the NAT causing problem? I am using static IP on both server and phones and do not really need NAT. If it is, how can I disable NAT on the FS server. Can anybody tell me the cause? Gary___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP documentation, interop, etc
I added a couple of things to that page for you. http://wiki.freeswitch.org/wiki/Specsheet Here is our interop list http://wiki.freeswitch.org/wiki/Interop_List /b On Oct 10, 2008, at 11:19 AM, Kristian Kielhofner wrote: Hello Freeswitch users, I've started working on a wiki for SIP documentation for interop and other features. I've created a basic page for Freeswitch: http://www.submityoursip.com/wiki/Freeswitch Would someone with more Freeswitch SIP experience help me fill in some more details? Things like support for various specs, issues when working with other devices, etc. I'm still trying to work on a template for implementations and some sort of format but I figured I'd get some information up and see where it goes. Thanks, and let me know if you have any suggestions! P.S. - Obviously it's not just limited to Freeswitch... If you have experience with other SIP implementations, service providers, etc I'd love to read about it! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with first use.
Can you join us on #freeswitch on irc.freenode.net? Press F8 and try again and see what it does you might also wanna try SVN Trunk. /b On Oct 10, 2008, at 11:20 AM, gary wrote: Hello, I am new to freeswitch. Just installed freeswitch-1.0.1 with default configurations on CentOS 4.6 with OpenVZ. I registered two Cisco7960 phones using static IP with default users(1001.xml, 1007.xml). The registration are OK, but when I try to call from each other, it immedially send to voicemail. The following is the log info from console: (I replaced the real IP with 10.1.1.117 in the log) 2008-10-10 11:41:02 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/ [EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes [bfe75310-638d-48df-8f3d-262143c15b22] 2008-10-10 11:41:02 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/[EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2008-10-10 11:41:02 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/ [EMAIL PROTECTED]:51029;transport=udp;fs_nat=yes) Ended 2008-10-10 11:41:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] :51029;transport=udp;fs_nat=yes [CS_HANGUP] Is it possible the NAT causing problem? I am using static IP on both server and phones and do not really need NAT. If it is, how can I disable NAT on the FS server. Can anybody tell me the cause? Gary ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with first use.
Here is the debug info after press F8:
2008-10-10 13:33:00 [DEBUG] sofia.c:3134 sofia_handle_sip_i_invite() IP
10.1.1.129 Rejected by acl domains. Falling back to Digest auth.
2008-10-10 13:33:00 [DEBUG] sofia.c:3134 sofia_handle_sip_i_invite() IP
10.1.1.129 Rejected by acl domains. Falling back to Digest auth.
2008-10-10 13:33:00 [NOTICE] switch_channel.c:534 switch_channel_set_name() New
Channel sofia/internal/[EMAIL PROTECTED] [41067ebd-0e4a-46e2-942c-f2a953819333]
2008-10-10 13:33:00 [DEBUG] sofia.c:3624 sofia_handle_sip_i_invite() Setting
NAT mode based on via port
2008-10-10 13:33:00 [DEBUG] sofia.c:2129 sofia_handle_sip_i_state() Channel
sofia/internal/[EMAIL PROTECTED] entering state [received]
2008-10-10 13:33:00 [DEBUG] sofia.c:2133 sofia_handle_sip_i_state() Remote SDP:
v=0
o=Cisco-SIPUA 11819 0 IN IP4 10.1.1.129
s=SIP Call
t=0 0
m=audio 23142 RTP/AVP 0 8 18 101
c=IN IP4 10.1.1.129
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2008-10-10 13:33:00 [DEBUG] sofia_glue.c:2280 sofia_glue_negotiate_sdp() Audio
Codec Compare [PCMU:0:8000]/[G722:9:8000]
2008-10-10 13:33:00 [DEBUG] sofia_glue.c:2280 sofia_glue_negotiate_sdp() Audio
Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
2008-10-10 13:33:00 [DEBUG] sofia_glue.c:1530 sofia_glue_tech_set_codec() Set
Codec sofia/internal/[EMAIL PROTECTED] PCMU/8000 20 ms 160 samples
2008-10-10 13:33:00 [DEBUG] sofia_glue.c:2243 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2008-10-10 13:33:00 [DEBUG] sofia.c:2270 sofia_handle_sip_i_state()
sofia/internal/[EMAIL PROTECTED] State Change CS_NEW - CS_INIT
2008-10-10 13:33:00 [DEBUG] switch_core_session.c:722
switch_core_session_signal_state_change() Kill sofia/internal/[EMAIL PROTECTED]
[BREAK]
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/internal/[EMAIL PROTECTED] Running State Change
CS_INIT
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State INIT
2008-10-10 13:33:00 [DEBUG] mod_sofia.c:80 sofia_on_init()
sofia/internal/[EMAIL PROTECTED] SOFIA INIT
2008-10-10 13:33:00 [DEBUG] mod_sofia.c:107 sofia_on_init()
sofia/internal/[EMAIL PROTECTED] State Change CS_INIT - CS_ROUTING
2008-10-10 13:33:00 [DEBUG] switch_core_session.c:722
switch_core_session_signal_state_change() Kill sofia/internal/[EMAIL PROTECTED]
[BREAK]
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:415
switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State INIT going
to sleep
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:365
switch_core_session_run() sofia/internal/[EMAIL PROTECTED] Running State Change
CS_ROUTING
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:420
switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State ROUTING
2008-10-10 13:33:00 [DEBUG] mod_sofia.c:119 sofia_on_routing()
sofia/internal/[EMAIL PROTECTED] SOFIA ROUTING
2008-10-10 13:33:00 [DEBUG] switch_core_state_machine.c:64
switch_core_standard_on_routing() Standard ROUTING sofia/internal/[EMAIL
PROTECTED]
2008-10-10 13:33:00 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing
Line1-[EMAIL PROTECTED]
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${unroll_loops}(true) =~ /^true$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${sip_looped_call}() =~ /^true$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${strftime(%H%M)}(1333) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions destination_number(1001) =~ /^886$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions destination_number(1001) =~ /^\*\*(\d+)$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions destination_number(1001) =~ /^870$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${network_addr}(10.1.1.129) =~ /^$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${numbering_plan}() =~ /^$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${call_debug}(false) =~ /^true$/
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:109 parse_exten() Regex mismatch
2008-10-10 13:33:00 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test
conditions ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
2008-10-10
[Freeswitch-users] Newb question on dialplan config
Hello, I am a FreeSwitch newb but have been using asterisk for a while now. I have a project for which I think FreeSwitch will be the best answer, so I need to learn. Have been reading the docs and followed the example at: http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk when I call from a Polycom on the asterisk box to a polycom on the freeswitch box all is good. When id do the reverse I.E. call the ast polycom from the freeswitch polycom I get only the following in the freswitch CLI: 2008-10-10 13:33:24 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [521c96a2-5205-bf46-9f9f-31124757b0ef] 2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing John Millican-2002 in context default 2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer sofia/internal/[EMAIL PROTECTED] to [EMAIL PROTECTED] 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-10 13:33:24 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/[EMAIL PROTECTED] [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 12 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] It would seem that the line: 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting is telling me my problem but I do not yet know why freeswitch does not have a route. I am certain that I have not correctly set the dial plan but haven't a clue what to look at. Both machines are on the 192.168.100.0 net, firewall is off on both the freeswitch box which is running on a VMware installation of WinXP SP3 and the asterisk box. I am using the default configs with the additions per the above page. I did have to change the following from the defaults in vars.xml to get 2 way audio when I call from asterisk to freeswitch: X-PRE-PROCESS cmd=set data=bind_server_ip=192.168.100.16/ X-PRE-PROCESS cmd=set data=external_rtp_ip=192.168.100.16/ X-PRE-PROCESS cmd=set data=external_sip_ip=192.168.100.16/ Any ideas? Is there something else I need to post to help decipher what I have done wrong or have not yet done? Thanks, JohnM ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] is tone_detect the right app?
I am trying to catch a key being pressed during a bridged call. The key could be pressed by either leg of the call. When the key is pressed, I want to play into both channels some sound file or send in some TTS output. Then after the playback is done, allow the callers to resume their conversation. For example, when someone in the call presses the 5 key, we want to say their account balance to both parties by doing some TTS or playback some wav files with the amount. But the parties may press the 5 key at any time during the call and we want to be able to detect it and react to it. Is Tone_Detect the right tool for this type of feature ? or am I missing the right one? If so, What frequency would we use for particular keys? Will tone_Detect sniff both legs or would we just do both r and w on the called leg? Can the timeout just be a very large number or can we leave out the timeout value so there is no timeout? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newb question on dialplan config
Brian West wrote:
Its looking for extension 2002 in context default on FreeSWITCH and
one doesn't exist so you get the NO ROUTE message.
Add a route to map 2002 so that it points at the Asterisk box.
/b
On Oct 10, 2008, at 1:00 PM, John Millican wrote:
Processing John Millican-2002 in context defau
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/external/[EMAIL PROTECTED]/
action application=hangup/
/condition
/extension
Is this not a routemap?
I apologize for such simple questions, but I am learning.
JohnM
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Re: [Freeswitch-users] Newb question on dialplan config
I'm going to guess you added it at the very bottom of the default.xml?
It needs to be above this line:
X-PRE-PROCESS cmd=include data=default/*.xml/
/b
On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/external/
[EMAIL PROTECTED]/
action application=hangup/
/condition
/extension
Is this not a routemap?
I apologize for such simple questions, but I am learning.
JohnM
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Re: [Freeswitch-users] SIP documentation, interop, etc
I've started working on a wiki for SIP documentation for interop and other features. I've created a basic page for Freeswitch: Kristian, We know you are an active member of the Asterisk community so we thank you for showing FS a little love! We appreciate it when OSS telephony users go the extra mile for the benefit of OSST as a whole. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newb question on dialplan config
Yep that was it! Now I just need to add a matching gateway, as I am
getting the error no matching gateway found, which I think I can
figure out.
Thank you for such quick and accurate help.
JohnM
Brian West wrote:
I'm going to guess you added it at the very bottom of the default.xml?
It needs to be above this line:
X-PRE-PROCESS cmd=include data=default/*.xml/
/b
On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default:
extension name=ast_extens
condition field=destination_number expression=^(2\d{3})$
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/external/[EMAIL PROTECTED]
mailto:sofia/external/[EMAIL PROTECTED]/
action application=hangup/
/condition
/extension
Is this not a routemap?
I apologize for such simple questions, but I am learning.
JohnM
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Re: [Freeswitch-users] SIP documentation, interop, etc
On 10/10/08, Brian West [EMAIL PROTECTED] wrote: I added a couple of things to that page for you. http://wiki.freeswitch.org/wiki/Specsheet Here is our interop list http://wiki.freeswitch.org/wiki/Interop_List /b Thanks Brian! I like your interop list. I wish more vendors/projects/etc would provide one. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Accepting SIP Calls from unregistered devices
Hi, How do i configure my Freeswitch to accept SIP calls from peers/devices not registered with it? Thanks, Klaus. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newb question on dialplan config
The new default configs in SVN trunk have a HUGE warning at the very bottom along with more documentation. I highly recommend you check it out. http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=9935 /b On Oct 10, 2008, at 1:46 PM, John Millican wrote: Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I think I can figure out. Thank you for such quick and accurate help. JohnM ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Anonymous or via IP auth? /b On Oct 10, 2008, at 1:48 PM, Klaus Teller wrote: Hi, How do i configure my Freeswitch to accept SIP calls from peers/ devices not registered with it? Thanks, Klaus. -- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Anonynous would be enough for me. Klaus. Original-Nachricht Datum: Fri, 10 Oct 2008 13:51:21 -0500 Von: Brian West [EMAIL PROTECTED] An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices Anonymous or via IP auth? /b On Oct 10, 2008, at 1:48 PM, Klaus Teller wrote: Hi, How do i configure my Freeswitch to accept SIP calls from peers/ devices not registered with it? Thanks, Klaus. -- ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- GMX Kostenlose Spiele: Einfach online spielen und Spaß haben mit Pastry Passion! http://games.entertainment.gmx.net/de/entertainment/games/free/puzzle/6169196 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
You have to do this... set auth-calls to false.
!-- everything above this is public --
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
!-- everything below this requires auth --
Just wedge this extension in your dialplan between the stuff you want
public and the stuff you want auth on. Then make sure you turn auth-
calls to false on your profile.
/b
On Oct 10, 2008, at 1:57 PM, Klaus Teller wrote:
Anonynous would be enough for me.
Klaus.
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Re: [Freeswitch-users] Newb question on dialplan config
Thanks, will do that right now. JohnM Brian West wrote: The new default configs in SVN trunk have a HUGE warning at the very bottom along with more documentation. I highly recommend you check it out. http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=9935 /b On Oct 10, 2008, at 1:46 PM, John Millican wrote: Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I think I can figure out. Thank you for such quick and accurate help. JohnM ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- John Millican Director of Technology Sentinel Communications, LLC PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 Fax (603) 764-7213 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Hmm, i'm not getting it right. Here is what i did:
1) In profile internal.xml i replaced
param name=auth-calls value=true/ with param name=auth-calls
value=false/
2) In profile external.xml this was already false.
3) Then i added the extension you provided both in public.xml and in
default.xml. In public.xml the extension is pasted just before the following
line:
!--
You can place files in the public directory to get included.
--
X-PRE-PROCESS cmd=include data=public/*.xml/
In default.xml it is placed just before the similar instructions.
Yet, i still get the following ACL error message from Freeswitch:
IP 192.168.2.34 Rejected by acl domains. My dialpaln and configuration is
almost the same i got from trunk one week ago.
I'm sure i'm doing few things wrong here. but what?
Thanks,
Klaus.
Original-Nachricht
Datum: Fri, 10 Oct 2008 14:01:57 -0500
Von: Brian West [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
You have to do this... set auth-calls to false.
!-- everything above this is public --
extension name=check_auth continue=true
condition field=${sip_authorized} expression=^true$
break=never
anti-action application=respond data=407/
/condition
/extension
!-- everything below this requires auth --
Just wedge this extension in your dialplan between the stuff you want
public and the stuff you want auth on. Then make sure you turn auth-
calls to false on your profile.
/b
On Oct 10, 2008, at 1:57 PM, Klaus Teller wrote:
Anonynous would be enough for me.
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen!
Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED]
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Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Oh yah remove the domains ACL param too it'll trump this :P /b On Oct 10, 2008, at 2:23 PM, Klaus Teller wrote: IP 192.168.2.34 Rejected by acl domains. My dialpaln and configuration is almost the same i got from trunk one week ago. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Well removing it off the sofia profile is optimal since you're going
to mix half open half authed.
/b
On Oct 10, 2008, at 2:45 PM, Klaus Teller wrote:
Great, thanks for the support. I've changed acl.conf.xml in the
following way:
list name=domains default=deny
node type=allow domain=$${domain}/
/list
into
list name=domains default=allow
node type=allow domain=$${domain}/
/list
Regards,
Klaus.
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Re: [Freeswitch-users] Open g729 g723 codec, any expierence
On 9/4/08, Anthony Minessale [EMAIL PROTECTED] wrote: We are going to produce a method to buy g729 for FreeSWITCH in the near future. Digging up an old thread... Any update here? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Open g729 g723 codec, any expierence
we are almost there. =D On Fri, Oct 10, 2008 at 3:06 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 9/4/08, Anthony Minessale [EMAIL PROTECTED] wrote: We are going to produce a method to buy g729 for FreeSWITCH in the near future. Digging up an old thread... Any update here? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Ringback with originate?
Hi,
I'm writing a script to connect one of my SIP phones with an external
number via a web interface. (Like click to call.) Right now, I'm
using xml_rpc and after some help from the guys on IRC, I've got it
working nicely.
When I execute the script, first my local SIP phone rings. I answer
it and hear silence until the person on the other end of the call
answers their phone.
It would be nice if I could simulate some kind of ringing sound so
that It sounds like I'm hearing the other phone ring like a normally
placed call.
I THOUGHT that the ringback variable would do this, but no matter
what I try, I don't get any sound.
Here is what I'm sending now...
valuestring{$${us-ring},accountcode=1}sofia/internal/
3235551212%111.111.111.111 amp;bridge(accountcode=1}sofia/internal/[EMAIL
PROTECTED]
) XML internal 3235551212 3235551212/string/value
Interestingly, (or not), I can control which phone works by the order
I use in the originate command. If I set MY phone first and the
outside number in the bridge, then I ring first. If I set the outside
number first and my phone in the bridge, then they ring first.
Any suggestions?
-Noah
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Re: [Freeswitch-users] Open g729 g723 codec, any expierence
On 10/10/08, Anthony Minessale [EMAIL PROTECTED] wrote: we are almost there. =D I'm sure I have a fairly steep learning curve in front of me but this has been the only issue holding me back from some serious Freeswitch fun. It's tough for me to even justify the time experimenting unless it's a viable platform in my environment (which means the ability to transcode G729). Can I at least be a beta tester or something? Please? I'm desperate!!! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Ok I think you and I aren't on the same page... can you clarify what you mean by my call to be transferred via event socket? /b On Oct 10, 2008, at 3:37 PM, Klaus Teller wrote: OK, i moved it to the sofia profile and it works just fine. Next step in my authorization problem. I call in with a softphone and want my call to be transferred via event socket to a remote application. Yet, Freeswitch rejects the call with the reason: [407][Proxy Authentication Required] session:sofia/internal/[EMAIL PROTECTED] I've got the outbound event socket working fine when i call from a peer registered with Freeswitch. But in this new scenario (with a non-registered device calling) Freeswitch is not happy. Anything i could change to make it work? Thanks, Klaus. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
Well, i want to manage calls remotely using the socket interface (as described here http://wiki.freeswitch.org/wiki/Event_socket_outbound). The calls i want to manage come from non-registered devices. Does that make sense? Klaus. Original-Nachricht Datum: Fri, 10 Oct 2008 15:44:09 -0500 Von: Brian West [EMAIL PROTECTED] An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices Ok I think you and I aren't on the same page... can you clarify what you mean by my call to be transferred via event socket? /b On Oct 10, 2008, at 3:37 PM, Klaus Teller wrote: OK, i moved it to the sofia profile and it works just fine. Next step in my authorization problem. I call in with a softphone and want my call to be transferred via event socket to a remote application. Yet, Freeswitch rejects the call with the reason: [407][Proxy Authentication Required] session:sofia/internal/[EMAIL PROTECTED] I've got the outbound event socket working fine when i call from a peer registered with Freeswitch. But in this new scenario (with a non-registered device calling) Freeswitch is not happy. Anything i could change to make it work? Thanks, Klaus. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
You have to route them to the park app. THen you can command and control them. /b On Oct 10, 2008, at 3:56 PM, Klaus Teller wrote: Well, i want to manage calls remotely using the socket interface (as described here http://wiki.freeswitch.org/wiki/ Event_socket_outbound). The calls i want to manage come from non- registered devices. Does that make sense? Klaus. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices
There are two things that i think are worth mentioning: 1) when i call from a peer registered to Freeswitch, my remote server is indeed notified and can do different thinks on the call. 2) I can also call registered peers using non-registered peers (thanks to the ACL settings we achieved previously) But the non-registerepeer cannot call extensions that are redirected to the event socket. I will try praking the call and let you know. I was just wondering why the callf rom registered peers don't need to be parked. Thanks, Klaus. Original-Nachricht Datum: Fri, 10 Oct 2008 16:00:18 -0500 Von: Brian West [EMAIL PROTECTED] An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices You have to route them to the park app. THen you can command and control them. /b On Oct 10, 2008, at 3:56 PM, Klaus Teller wrote: Well, i want to manage calls remotely using the socket interface (as described here http://wiki.freeswitch.org/wiki/ Event_socket_outbound). The calls i want to manage come from non- registered devices. Does that make sense? Klaus. -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Open g729 g723 codec, any expierence
I would love to be a beta tester too! I haven't switched from Asterisk for the same reason. Cheers, Nicolas On Fri, Oct 10, 2008 at 4:38 PM, Michael Collins [EMAIL PROTECTED] wrote: Can I at least be a beta tester or something? Please? I'm desperate!!! Dude, you're hired! :) -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Open g729 g723 codec, any expierence
Gee, me too Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), Http://www.enterux.com/ On 11-Oct-08, at 4:43, Nicolas Brenner [EMAIL PROTECTED] wrote: I would love to be a beta tester too! I haven't switched from Asterisk for the same reason. Cheers, Nicolas On Fri, Oct 10, 2008 at 4:38 PM, Michael Collins [EMAIL PROTECTED] wrote: Can I at least be a beta tester or something? Please? I'm desperate!!! Dude, you're hired! :) -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Open g729 g723 codec, any expierence
As there seems to be quite some interest in G.729, those interested who can commit to purchasing licenses, please update this wiki page: http://wiki.freeswitch.org/wiki/Bounty#G729_Licensing_Bounty Mike On Oct 10, 2008, at 11:55 PM, Alex Vostrikov wrote: Mitul Limbani wrote: yeah yeah, mee too, guys, please!!! Gee, me too Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), Http://www.enterux.com/ On 11-Oct-08, at 4:43, Nicolas Brenner[EMAIL PROTECTED] wrote: I would love to be a beta tester too! I haven't switched from Asterisk for the same reason. Cheers, Nicolas On Fri, Oct 10, 2008 at 4:38 PM, Michael Collins[EMAIL PROTECTED] wrote: Can I at least be a beta tester or something? Please? I'm desperate!!! Dude, you're hired! :) -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
