Hi all! I'm new to FreeSwitch and have a newby question: what is the best/preferred way to create a new call from some external application? For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ... I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call... Thanks in advance, Boris
One option is to use XML-RPC and originate a call. http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC You can issue an "api originate" from an XML-RPC request with ease. /b On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@freeswitch.org
Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC? Thanks in advance, Boris On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote:
One option is to use XML-RPC and originate a call.
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
You can issue an "api originate" from an XML-RPC request with ease.
/b
On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
uuid_send_dtmf <uuid> <dtmf_data> Could somone please wiki this, it seems to not be documented. Mike On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote:
Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote: One option is to use XML-RPC and originate a call.
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
You can issue an "api originate" from an XML-RPC request with ease.
/b
On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@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
I think I heard my name! I will put this down as an item for tomorrow's call. -MC ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, July 22, 2008 2:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] outbound calling uuid_send_dtmf <uuid> <dtmf_data> Could somone please wiki this, it seems to not be documented. Mike On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote: Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC? Thanks in advance, Boris On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote: One option is to use XML-RPC and originate a call. http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC You can issue an "api originate" from an XML-RPC request with ease. /b On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@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
In the default context I have <extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/nat/1001%24.67.78.200" /> </condition> </extension I am trying to call the 1001 extension and am getting 2008-07-23 11:26:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Jair->1001@default 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1000.2008-07-23-11-26-13.wa v 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-07-23 11:26:13 [WARNING] switch_ivr_originate.c:539 switch_ivr_originate() No origination URL specified! 2008-07-23 11:26:13 [ERR] switch_ivr_originate.c:823 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [DESTINATION_OUT_OF_ORDER] What am I missing ? thanks Jair Santos
Users don't register to the nat profile at all and SHOULDNOT. The default config makes the assumption that all users are registering to the internal profile. Anything outside of that would require you to know a bit more on how FreeSWITCH works. Join us on IRC #freeswitch on irc.freenode.net and people can help you get up to speed. /b On Jul 23, 2008, at 1:33 PM, Jair Santos wrote:
In the default context I have
<extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/nat/1001%24.67.78.200" /> </condition> </extension
I am trying to call the 1001 extension and am getting
2008-07-23 11:26:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Jair->1001@default 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ usr/local/freeswitch/recordings/1000.2008-07-23-11-26-13.wav 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-07-23 11:26:13 [WARNING] switch_ivr_originate.c:539 switch_ivr_originate() No origination URL specified! 2008-07-23 11:26:13 [ERR] switch_ivr_originate.c:823 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [DESTINATION_OUT_OF_ORDER]
What am I missing ?
thanks
Jair Santos
Brian West sip:brian@freeswitch.org
Changing to <extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/internal/1001%24.67.78.200" /> </condition> </extension I get 2008-07-23 11:49:10 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/internal/1000@192.168.1.117 [82981f3f-de3f-4acf-bc5e-d49e6595e0aa] 2008-07-23 11:49:10 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Jair->1001@default 2008-07-23 11:49:10 [WARNING] mod_sofia.c:1807 sofia_outgoing_channel() Cannot locate registered user 1001@24.67.78.200 But the user is registered as I can see in the phone. Jair Santos -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 11:43 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling Users don't register to the nat profile at all and SHOULDNOT. The default config makes the assumption that all users are registering to the internal profile. Anything outside of that would require you to know a bit more on how FreeSWITCH works. Join us on IRC #freeswitch on irc.freenode.net and people can help you get up to speed. /b On Jul 23, 2008, at 1:33 PM, Jair Santos wrote: In the default context I have <extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/nat/1001%24.67.78.200" /> </condition> </extension I am trying to call the 1001 extension and am getting 2008-07-23 11:26:13 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Jair->1001@default 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1000.2008-07-23-11-26-13.wa v 2008-07-23 11:26:13 [INFO] switch_ivr_async.c:1443 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-07-23 11:26:13 [WARNING] switch_ivr_originate.c:539 switch_ivr_originate() No origination URL specified! 2008-07-23 11:26:13 [ERR] switch_ivr_originate.c:823 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [DESTINATION_OUT_OF_ORDER] What am I missing ? thanks Jair Santos Brian West sip:brian@freeswitch.org
It's fairly obvious you don't have a user 1001 in domain 24.67.78.200, I also recommend you can update to the latest code as of now. Did you modify the default config? And what scenario are you trying to make this work with? On Jul 23, 2008, at 1:53 PM, Jair Santos wrote:
2008-07-23 11:49:10 [WARNING] mod_sofia.c:1807 sofia_outgoing_channel() Cannot locate registered user 1001@24.67.78.200
Brian West sip:brian@freeswitch.org
I have changed the user 1001 to <include> <user id="1001" domain name = "24.67.78.200" mailbox="1001"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1001"/> </params> <variables> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> </variables> </user> </include> I can see the 1001 user registered in the phone. Trying to phone->FS->nat->phone -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:07 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling It's fairly obvious you don't have a user 1001 in domain 24.67.78.200, I also recommend you can update to the latest code as of now. Did you modify the default config? And what scenario are you trying to make this work with? On Jul 23, 2008, at 1:53 PM, Jair Santos wrote: 2008-07-23 11:49:10 [WARNING] mod_sofia.c:1807 sofia_outgoing_channel() Cannot locate registered user 1001@24.67.78.200 Brian West sip:brian@freeswitch.org
API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED internal2 profile sip:mod_sofia@24.67.78.200:5061 RUNNING (0) external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED ============================================================================ ===================== -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling You don't add domain name in there like that. show me the output of "sofia status". /b On Jul 23, 2008, at 2:11 PM, Jair Santos wrote: <user id="1001" domain name = "24.67.78.200" mailbox="1001"> Brian West sip:brian@freeswitch.org
Your domain is 192.168.1.117 ... Is freeswitch behind nat? /b On Jul 23, 2008, at 2:21 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED internal2 profile sip:mod_sofia@24.67.78.200:5061 RUNNING (0) external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
You don't add domain name in there like that.
show me the output of "sofia status".
/b
On Jul 23, 2008, at 2:11 PM, Jair Santos wrote:
<user id="1001" domain name = "24.67.78.200" mailbox="1001">
Brian West sip:brian@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
Brian West sip:brian@freeswitch.org
Yes. And I want to register this user to the external IP . I am forwarding the port to the box. How can I register the user trough the ext IP ? Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
Your domain is 192.168.1.117 ... Is freeswitch behind nat?
/b
On Jul 23, 2008, at 2:21 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
internal profile
sip:mod_sofia@192.168.1.117:5060
RUNNING (0) nat profile
sip:mod_sofia@24.67.78.200:5070
RUNNING (0) default alias internal ALIASED internal2 profile
sip:mod_sofia@24.67.78.200:5061
RUNNING (0) external profile
sip:mod_sofia@24.67.78.200:5080
RUNNING (0) inphonex gateway
sip:3462101@sip.varphonex.com
REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
You don't add domain name in there like that.
show me the output of "sofia status".
/b
On Jul 23, 2008, at 2:11 PM, Jair Santos wrote:
<user id="1001" domain name = "24.67.78.200" mailbox="1001">
Brian West sip:brian@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/freesw itch-users
Brian West sip:brian@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/freesw itch-users http://www.freeswitch.org
Please follow http://wiki.freeswitch.org/wiki/Example_Offsite_phones I managed to get the SIP remote extensions work for the double NATed environment. Just need to be more careful. Thanks, Chris On Wed, Jul 23, 2008 at 3:59 PM, Jair Santos <jairds@shaw.ca> wrote:
Yes.
And I want to register this user to the external IP . I am forwarding the port to the box. How can I register the user trough the ext IP ?
Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
Your domain is 192.168.1.117 ... Is freeswitch behind nat?
/b
On Jul 23, 2008, at 2:21 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
internal profile
sip:mod_sofia@192.168.1.117:5060
RUNNING (0) nat profile
sip:mod_sofia@24.67.78.200:5070
RUNNING (0) default alias internal ALIASED internal2 profile
sip:mod_sofia@24.67.78.200:5061
RUNNING (0) external profile
sip:mod_sofia@24.67.78.200:5080
RUNNING (0) inphonex gateway
sip:3462101@sip.varphonex.com <sip%3A3462101@sip.varphonex.com>
REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
You don't add domain name in there like that.
show me the output of "sofia status".
/b
On Jul 23, 2008, at 2:11 PM, Jair Santos wrote:
<user id="1001" domain name = "24.67.78.200" mailbox="1001">
Brian West sip:brian@freeswitch.org <sip%3Abrian@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/freesw itch-users
Brian West sip:brian@freeswitch.org <sip%3Abrian@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/freesw itch-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
Thank you. I have followed the instructions of this link but FS can't find the user. I've stopped IPtables and have forwarded the port 5090 to the box. sofia status ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED internal2 profile sip:mod_sofia@24.67.78.200:5061 RUNNING (0) external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED doublenat profile sip:mod_sofia@24.67.78.200:5090 RUNNING (0) ============================================================================ ===================== Jair -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of unknown Sent: Wednesday, July 23, 2008 1:06 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling Please follow http://wiki.freeswitch.org/wiki/Example_Offsite_phones I managed to get the SIP remote extensions work for the double NATed environment. Just need to be more careful. Thanks, Chris On Wed, Jul 23, 2008 at 3:59 PM, Jair Santos <jairds@shaw.ca> wrote: Yes. And I want to register this user to the external IP . I am forwarding the port to the box. How can I register the user trough the ext IP ? Jair Santos
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, July 23, 2008 12:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
Your domain is 192.168.1.117 ... Is freeswitch behind nat?
/b
On Jul 23, 2008, at 2:21 PM, Jair Santos wrote:
API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
internal profile
sip:mod_sofia@192.168.1.117:5060
RUNNING (0) nat profile
sip:mod_sofia@24.67.78.200:5070
RUNNING (0) default alias internal ALIASED internal2 profile
sip:mod_sofia@24.67.78.200:5061
RUNNING (0) external profile
sip:mod_sofia@24.67.78.200:5080
RUNNING (0) inphonex gateway
sip:3462101@sip.varphonex.com <mailto:sip%3A3462101@sip.varphonex.com>
REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = =
======================================================================
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 12:17 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling
You don't add domain name in there like that.
show me the output of "sofia status".
/b
On Jul 23, 2008, at 2:11 PM, Jair Santos wrote:
<user id="1001" domain name = "24.67.78.200" mailbox="1001">
Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
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Brian West sip:brian@freeswitch.org <mailto:sip%3Abrian@freeswitch.org>
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Oh btw press F8 if you're on linux. The default config has 1000 thru 1019 configured already so if you added that at the bottom of default then the Local_Extensions is overriding it. /b On Jul 23, 2008, at 1:33 PM, Jair Santos wrote:
<extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/nat/1001%24.67.78.200" /> </condition> </extension
Brian West sip:brian@freeswitch.org
I added to the top. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, July 23, 2008 11:45 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] registered user calling Oh btw press F8 if you're on linux. The default config has 1000 thru 1019 configured already so if you added that at the bottom of default then the Local_Extensions is overriding it. /b On Jul 23, 2008, at 1:33 PM, Jair Santos wrote: <extension name="internal"> <condition field="destination_number" expression="^1001"/> <action application="bridge" data="sofia/nat/1001%24.67.78.200" /> </condition> </extension Brian West sip:brian@freeswitch.org
That was fast :) I'm struggling with the FreeSwitch documentation available on the net and cannot tie all parts together to see the big picture for what I'm trying to achieve: >From an external application drive operations like: * originate a call to a call leg * feed dtmf to call leg * collect dtmf from a call leg * play an announcement to a call leg * join two call legs into a call * ... And for all above mentioned actions I would like to be notified about status of action, i.e. when an action is completed and its status (successful, failed, ...). I was playing with the XML-RPC, but seems that actions like uuid_send_dtmf and uuid_broadcast are asynchronous and therefore client is not able to detect its completition. Is there something I'm missing here? Is the XML-RPC approach suitable for what I'm trying to accieve? Thanks in advance, Boris On Tue, Jul 22, 2008 at 11:55 PM, Michael Jerris <mike@jerris.com> wrote: > uuid_send_dtmf <uuid> <dtmf_data> > Could somone please wiki this, it seems to not be documented. > > Mike > > On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote: > > Thanks for pointing me into the right direction. I manager to create a call > from Java app without any problems using XML-RPC. Btw., is there a way to > generate dtmf digits to a call leg using XML-RPC? > > Thanks in advance, > Boris > > On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote: > >> One option is to use XML-RPC and originate a call. >> >> http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC >> >> You can issue an "api originate" from an XML-RPC request with ease. >> >> /b >> >> On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote: >> >> > Hi all! >> > >> > I'm new to FreeSwitch and have a newby question: what is the best/ >> > preferred way to create a new call from some external application? >> > >> > For example: I have an application A (i.e. stand alone Java >> > application) which will invoke a RMI/web service or whatever service >> > which should reside within FreeSwitch and will, when called, create >> > a new call leg, play an audio message to this call leg, ... >> > >> > I'm currently browsing some samples on how to write a Java >> > application that resides within FreeSwitch using mod_java. Is this >> > the right way to do it? The problem is that I want to create a call >> > leg from scratch and not to handle an incoming call and do something >> > with it; all samples only process an incoming call... >> > >> > Thanks in advance, >> > Boris >> >> Brian West >> sip:brian@freeswitch.org <sip%3Abrian@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 > > > > _______________________________________________ > 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 > >
You might be better off using Mod_event_socket and some lua. Check out mod_lua and mod_event_socket on the wiki. /b On Jul 23, 2008, at 4:13 AM, Boris Krivonog wrote:
That was fast :)
I'm struggling with the FreeSwitch documentation available on the net and cannot tie all parts together to see the big picture for what I'm trying to achieve:
From an external application drive operations like: * originate a call to a call leg * feed dtmf to call leg * collect dtmf from a call leg * play an announcement to a call leg * join two call legs into a call * ...
And for all above mentioned actions I would like to be notified about status of action, i.e. when an action is completed and its status (successful, failed, ...). I was playing with the XML-RPC, but seems that actions like uuid_send_dtmf and uuid_broadcast are asynchronous and therefore client is not able to detect its completition. Is there something I'm missing here? Is the XML-RPC approach suitable for what I'm trying to accieve?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 11:55 PM, Michael Jerris <mike@jerris.com> wrote: uuid_send_dtmf <uuid> <dtmf_data>
Could somone please wiki this, it seems to not be documented.
Mike
On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote:
Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote: One option is to use XML-RPC and originate a call.
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
You can issue an "api originate" from an XML-RPC request with ease.
/b
On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@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
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Brian West sip:brian@freeswitch.org
you will need event socket for what you want to do. On Wed, Jul 23, 2008 at 4:13 AM, Boris Krivonog <boris.krivonog@gmail.com> wrote:
That was fast :)
I'm struggling with the FreeSwitch documentation available on the net and cannot tie all parts together to see the big picture for what I'm trying to achieve:
From an external application drive operations like: * originate a call to a call leg * feed dtmf to call leg * collect dtmf from a call leg * play an announcement to a call leg * join two call legs into a call * ...
And for all above mentioned actions I would like to be notified about status of action, i.e. when an action is completed and its status (successful, failed, ...). I was playing with the XML-RPC, but seems that actions like uuid_send_dtmf and uuid_broadcast are asynchronous and therefore client is not able to detect its completition. Is there something I'm missing here? Is the XML-RPC approach suitable for what I'm trying to accieve?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 11:55 PM, Michael Jerris <mike@jerris.com> wrote:
uuid_send_dtmf <uuid> <dtmf_data> Could somone please wiki this, it seems to not be documented.
Mike
On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote:
Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote:
One option is to use XML-RPC and originate a call.
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
You can issue an "api originate" from an XML-RPC request with ease.
/b
On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks for your info, seems event sockets are the thing for me :) I was playing with FreeSwitch and event sockets for a few days now and I must confess I'm very impressed with the ease and power of FreeSwitch! Again, thanks for help, Boris On Wed, Jul 23, 2008 at 3:14 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
you will need event socket for what you want to do.
On Wed, Jul 23, 2008 at 4:13 AM, Boris Krivonog <boris.krivonog@gmail.com> wrote:
That was fast :)
I'm struggling with the FreeSwitch documentation available on the net and cannot tie all parts together to see the big picture for what I'm trying to achieve:
From an external application drive operations like: * originate a call to a call leg * feed dtmf to call leg * collect dtmf from a call leg * play an announcement to a call leg * join two call legs into a call * ...
And for all above mentioned actions I would like to be notified about status of action, i.e. when an action is completed and its status (successful, failed, ...). I was playing with the XML-RPC, but seems that actions like uuid_send_dtmf and uuid_broadcast are asynchronous and therefore client is not able to detect its completition. Is there something I'm missing here? Is the XML-RPC approach suitable for what I'm trying to accieve?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 11:55 PM, Michael Jerris <mike@jerris.com> wrote:
uuid_send_dtmf <uuid> <dtmf_data> Could somone please wiki this, it seems to not be documented.
Mike
On Jul 22, 2008, at 5:52 PM, Boris Krivonog wrote:
Thanks for pointing me into the right direction. I manager to create a call from Java app without any problems using XML-RPC. Btw., is there a way to generate dtmf digits to a call leg using XML-RPC?
Thanks in advance, Boris
On Tue, Jul 22, 2008 at 8:53 PM, Brian West <brian@freeswitch.org> wrote:
One option is to use XML-RPC and originate a call.
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
You can issue an "api originate" from an XML-RPC request with ease.
/b
On Jul 22, 2008, at 1:14 PM, Boris Krivonog wrote:
Hi all!
I'm new to FreeSwitch and have a newby question: what is the best/ preferred way to create a new call from some external application?
For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ...
I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call...
Thanks in advance, Boris
Brian West sip:brian@freeswitch.org <sip%3Abrian@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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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participants (7)
-
Anthony Minessale
-
Boris Krivonog
-
Brian West
-
Jair Santos
-
Michael Collins
-
Michael Jerris
-
unknown