Re: [Freeswitch-users] Call not originated through event socket
Hi Steven, Thanks for responding. I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine. Thanks. -Ashish On Fri, Apr 5, 2013 at 12:17 AM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) 2. Re: freeswitch say application currency in multipal language (Michael Collins) 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) 4. Re: Call not originated through event socket (Steven Ayre)
---------- Forwarded message ---------- From: Vik Killa <vipkilla@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 14:13:26 -0400 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Please let us know if and how you get it resolved. I'll do the same if I can figure it out. Thanks.
<snip>
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 11:14:52 -0700 Subject: Re: [Freeswitch-users] freeswitch say application currency in multipal language I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome.
Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas:
euro, euros franc, francs Canadian, Australian, US dollar/dollars pound, pounds
Send me some more ideas and I will get them added to the to-be-recorded list.
-MC
On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388@gmail.com>wrote:
Hi all, I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play.
Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible.
Is that any easy way to do this thing in multi language currency play in say application.
i use this syntax in my free-switch dial plan $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>";
Thanks In advance...
-- Thanks, Bhavik Patel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 21:16:04 +0300 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Indeed, outbound_caller_id_name is just a variable.
To use it you have to tell the dialplan to set effective_caller_id_name=${ outbound_caller_id_name}. effective_caller_id_name IS a variable that actually gets used. (The default dialplan does this.)
-Avi Marcus BestFone
On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < schoch+freeswitch.org@xwin32.com> wrote:
In my directory, I set the variable "effective_caller_id_name" instead of "internal_caller_id_name". (I also set the "outbound_caller_id_name" variable just like your example, although that never seems to get sent to the PSTN.)
-- Steve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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---------- Forwarded message ---------- From: Steven Ayre <steveayre@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 19:46:22 +0100 Subject: Re: [Freeswitch-users] Call not originated through event socket +1
'/log 9' and watch the debug output. It'll be very clear about what's happening. Most likely there's something wrong about your usage of originate.
-Steve
On 4 April 2013 19:08, Michael Collins <msc@freeswitch.org> wrote:
You could also just hop on fs_cli and watch the output while you make a test call with the event socket. Most likely you'll see what's going on.
-MC
On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < Hector.Geraldino@ipsoft.com> wrote:
+OK doesn’t mean the call was generated, it just means that FreeSWITCH received (and processed) the command.****
** **
You should probably be better served by listening to the events for each one of these calls. There are a few handful events that can tell you if the channel was created, the call was originated, if it was answered and when it was dropped with the drop reason.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ashish gautam *Sent:* Thursday, April 04, 2013 5:26 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Call not originated through event socket** **
** **
Hi,****
** **
I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated.****
** **
Any help is appreciated.****
** **
Thanks.****
** **
-Ashish****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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NORMAL_CIRCUIT_CONGESTION is an error that can be returned by the SIP server you're calling out through. Show us the debug log and enable sip trace (sofia global siptrace on). Almost certainly that error is coming from your SIP provider and you'll need to take it up with them to see why they're returning that. Check for the SIP code in the final response they return (eg 503 Service Unavailable) and the Reason header in that that message (which is where you may see NORMAL_CIRCUIT_CONGESTION). -Steve On 5 April 2013 07:24, Ashish gautam <ashish@nms.co.in> wrote:
Hi Steven,
Thanks for responding.
I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine.
Thanks. -Ashish
On Fri, Apr 5, 2013 at 12:17 AM, < freeswitch-users-request@lists.freeswitch.org> wrote:
Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org
To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org
You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org
When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:
1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) 2. Re: freeswitch say application currency in multipal language (Michael Collins) 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) 4. Re: Call not originated through event socket (Steven Ayre)
---------- Forwarded message ---------- From: Vik Killa <vipkilla@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 14:13:26 -0400 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Please let us know if and how you get it resolved. I'll do the same if I can figure it out. Thanks.
<snip>
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 11:14:52 -0700 Subject: Re: [Freeswitch-users] freeswitch say application currency in multipal language I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome.
Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas:
euro, euros franc, francs Canadian, Australian, US dollar/dollars pound, pounds
Send me some more ideas and I will get them added to the to-be-recorded list.
-MC
On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388@gmail.com>wrote:
Hi all, I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play.
Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible.
Is that any easy way to do this thing in multi language currency play in say application.
i use this syntax in my free-switch dial plan $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>";
Thanks In advance...
-- Thanks, Bhavik Patel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 21:16:04 +0300 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Indeed, outbound_caller_id_name is just a variable.
To use it you have to tell the dialplan to set effective_caller_id_name=${outbound_caller_id_name}. effective_caller_id_name IS a variable that actually gets used. (The default dialplan does this.)
-Avi Marcus BestFone
On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < schoch+freeswitch.org@xwin32.com> wrote:
In my directory, I set the variable "effective_caller_id_name" instead of "internal_caller_id_name". (I also set the "outbound_caller_id_name" variable just like your example, although that never seems to get sent to the PSTN.)
-- Steve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
---------- Forwarded message ---------- From: Steven Ayre <steveayre@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 19:46:22 +0100 Subject: Re: [Freeswitch-users] Call not originated through event socket +1
'/log 9' and watch the debug output. It'll be very clear about what's happening. Most likely there's something wrong about your usage of originate.
-Steve
On 4 April 2013 19:08, Michael Collins <msc@freeswitch.org> wrote:
You could also just hop on fs_cli and watch the output while you make a test call with the event socket. Most likely you'll see what's going on.
-MC
On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < Hector.Geraldino@ipsoft.com> wrote:
+OK doesn’t mean the call was generated, it just means that FreeSWITCH received (and processed) the command.****
** **
You should probably be better served by listening to the events for each one of these calls. There are a few handful events that can tell you if the channel was created, the call was originated, if it was answered and when it was dropped with the drop reason.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ashish gautam *Sent:* Thursday, April 04, 2013 5:26 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Call not originated through event socket* ***
** **
Hi,****
** **
I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated.****
** **
Any help is appreciated.****
** **
Thanks.****
** **
-Ashish****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Hi Steve, I am generating outbound calls to the PSTN network connected to my FreeSWITCH box via PRI line. Below is a piece of debug log for the same. There is some more but I am pasting the error lines out of that: 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:3/9891605603 ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:3/9891605603 [ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5] ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:3/9891605603) State Change CS_NEW -> CS_INIT ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:3/9891605603 [BREAK] 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 to channel 1:3 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c3][1:3] Changed state from DOWN to DIALING 2013-04-05 12:10:13.234533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[95] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9467797613] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] 2013-04-05 12:10:13.234533 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:4 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20ms 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:4/9467797613 ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:4/9467797613 [ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5] ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:4/9467797613) State Change CS_NEW -> CS_INIT ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:4/9467797613 [BREAK] 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 to channel 1:4 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c4][1:4] Changed state from DOWN to DIALING 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[96] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8802865008] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[97] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8826454579] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[98] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9868599566] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true] ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW] ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep 2013-04-05 12:10:13.274533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2013-04-05 12:10:13.274533 [DEBUG] ftdm_state.c:541 [s1c1][1:1] Executing state processor for DIALING 2013-04-05 12:10:13.274533 [DEBUG] ftmod_libpri.c:935 -- 1:1 STATE [DIALING] 2013-04-05 12:10:13.274533 [DEBUG] mod_freetdm.c:2657 got clear channel sig [DIALING] On Fri, Apr 5, 2013 at 12:33 PM, Steven Ayre <steveayre@gmail.com> wrote:
NORMAL_CIRCUIT_CONGESTION is an error that can be returned by the SIP server you're calling out through. Show us the debug log and enable sip trace (sofia global siptrace on).
Almost certainly that error is coming from your SIP provider and you'll need to take it up with them to see why they're returning that.
Check for the SIP code in the final response they return (eg 503 Service Unavailable) and the Reason header in that that message (which is where you may see NORMAL_CIRCUIT_CONGESTION).
-Steve
On 5 April 2013 07:24, Ashish gautam <ashish@nms.co.in> wrote:
Hi Steven,
Thanks for responding.
I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine.
Thanks. -Ashish
On Fri, Apr 5, 2013 at 12:17 AM, < freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) 2. Re: freeswitch say application currency in multipal language (Michael Collins) 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) 4. Re: Call not originated through event socket (Steven Ayre)
---------- Forwarded message ---------- From: Vik Killa <vipkilla@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 14:13:26 -0400 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Please let us know if and how you get it resolved. I'll do the same if I can figure it out. Thanks.
<snip>
---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 11:14:52 -0700 Subject: Re: [Freeswitch-users] freeswitch say application currency in multipal language I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome.
Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas:
euro, euros franc, francs Canadian, Australian, US dollar/dollars pound, pounds
Send me some more ideas and I will get them added to the to-be-recorded list.
-MC
On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388@gmail.com
wrote:
Hi all, I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play.
Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible.
Is that any easy way to do this thing in multi language currency play in say application.
i use this syntax in my free-switch dial plan $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>";
Thanks In advance...
-- Thanks, Bhavik Patel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
---------- Forwarded message ---------- From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 21:16:04 +0300 Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous! Indeed, outbound_caller_id_name is just a variable.
To use it you have to tell the dialplan to set effective_caller_id_name=${outbound_caller_id_name}. effective_caller_id_name IS a variable that actually gets used. (The default dialplan does this.)
-Avi Marcus BestFone
On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < schoch+freeswitch.org@xwin32.com> wrote:
In my directory, I set the variable "effective_caller_id_name" instead of "internal_caller_id_name". (I also set the "outbound_caller_id_name" variable just like your example, although that never seems to get sent to the PSTN.)
-- Steve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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---------- Forwarded message ---------- From: Steven Ayre <steveayre@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Thu, 4 Apr 2013 19:46:22 +0100 Subject: Re: [Freeswitch-users] Call not originated through event socket +1
'/log 9' and watch the debug output. It'll be very clear about what's happening. Most likely there's something wrong about your usage of originate.
-Steve
On 4 April 2013 19:08, Michael Collins <msc@freeswitch.org> wrote:
You could also just hop on fs_cli and watch the output while you make a test call with the event socket. Most likely you'll see what's going on.
-MC
On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < Hector.Geraldino@ipsoft.com> wrote:
+OK doesn’t mean the call was generated, it just means that FreeSWITCH received (and processed) the command.****
** **
You should probably be better served by listening to the events for each one of these calls. There are a few handful events that can tell you if the channel was created, the call was originated, if it was answered and when it was dropped with the drop reason.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ashish gautam *Sent:* Thursday, April 04, 2013 5:26 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Call not originated through event socket ****
** **
Hi,****
** **
I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated.****
** **
Any help is appreciated.****
** **
Thanks.****
** **
-Ashish****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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participants (2)
-
Ashish gautam
-
Steven Ayre