When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received. Here's the captured flow: |Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) | After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated. I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint. Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas. Thanks! Hynek
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue? Hynek On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
There are some use-cases. The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing. -Steve On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com> wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
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The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console. I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else. console loglevel debug sofia global siptrace on On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com> wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com> wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ. Hynek On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console.
I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else.
console loglevel debug sofia global siptrace on
On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com> wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com> wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
_________________________________________________________________________ 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
_________________________________________________________________________ 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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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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I think I see why, nobody seems to use those callstate events enough to notice anything wrong with them, Try latest HEAD. On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ.
Hynek
On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console.
I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else.
console loglevel debug sofia global siptrace on
On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com>wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com> wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
_________________________________________________________________________ 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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_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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With the HEAD it now works as expected, RINGING comes only after 180/183 is received from the originated endpoint. Could you also merge the change into more stable branch? Do you want me to file a bug for that? Hynek On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think I see why, nobody seems to use those callstate events enough to notice anything wrong with them, Try latest HEAD.
On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ.
Hynek
On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console.
I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else.
console loglevel debug sofia global siptrace on
On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com>wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com> wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar@gmail.com
wrote:
When originating a call the respective call channel's call state is set to RINGING right after progress 100 is received.
Here's the captured flow:
|Time | <src ip> | | | | <dst ip> | |2.848 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.848 | 407 Proxy Authentication Required |SIP Status | |(5080) <------------------ (5060) | |2.849 | ACK | |SIP Request | |(5080) ------------------> (5060) | |2.849 | INVITE SDP (g711A g711U GSM telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed To:<sip:endpoint removed | |(5080) ------------------> (5060) | |2.850 | 100 Trying| |SIP Status | |(5080) <------------------ (5060) | |13.444 | 180 Ringing |SIP Status | |(5080) <------------------ (5060) | |13.445 | 183 Session Progress SDP (g711A g711U GSM tele...ne-eventRTPType-101) |SIP Status | |(5080) <------------------ (5060) | |13.445 | RTP (g711A) |RTP Num packets:230 Duration:4.574s SSRC:0x1E777E26 | |(26056) <------------------ (19312) | |13.601 | RTP (g711A) |RTP Num packets:220 Duration:4.419s SSRC:0xA530C519 | |(26056) ------------------> (19312) |
After the 100 Trying is received the switch executes switch_channel_perform_set_running_state (switch_channel.c) and the channel call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is generated.
I would expect the channel call state to be set to RINGING only after 180 Ringing is received from the far endpoint.
Could anybody give me a hint what could be wrong or what steps to take to figure out? I am already out of ideas.
Thanks! Hynek
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
Head is the most stable branch currently, its in final stage. We will not merge things until 1.2 is branched and 1.5 is tagged on head. On Feb 23, 2013 3:51 PM, "Hynek Cihlar" <hynek.cihlar@gmail.com> wrote:
With the HEAD it now works as expected, RINGING comes only after 180/183 is received from the originated endpoint.
Could you also merge the change into more stable branch? Do you want me to file a bug for that?
Hynek
On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think I see why, nobody seems to use those callstate events enough to notice anything wrong with them, Try latest HEAD.
On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ.
Hynek
On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console.
I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else.
console loglevel debug sofia global siptrace on
On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com>wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
Is there even a valid use case for Freeswitch to set a channel to RINGING before the actual ring is signaled by the far endpoint? What other evidence would be helpful to diagnose the issue?
Hynek
On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar < hynek.cihlar@gmail.com> wrote:
> When originating a call the respective call channel's call state is > set to RINGING right after progress 100 is received. > > Here's the captured flow: > > |Time | <src ip> | > | | | <dst ip> | > > |2.848 | INVITE SDP (g711A g711U GSM > telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed > To:<sip:endpoint removed > | |(5080) ------------------> (5060) | > |2.848 | 407 Proxy Authentication Required |SIP > Status > | |(5080) <------------------ (5060) | > |2.849 | ACK | |SIP Request > | |(5080) ------------------> (5060) | > |2.849 | INVITE SDP (g711A g711U GSM > telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed > To:<sip:endpoint removed > | |(5080) ------------------> (5060) | > |2.850 | 100 Trying| |SIP Status > | |(5080) <------------------ (5060) | > |13.444 | 180 Ringing |SIP Status > | |(5080) <------------------ (5060) | > |13.445 | 183 Session Progress SDP (g711A g711U GSM > tele...ne-eventRTPType-101) |SIP Status > | |(5080) <------------------ (5060) | > |13.445 | RTP (g711A) |RTP Num > packets:230 Duration:4.574s SSRC:0x1E777E26 > | |(26056) <------------------ (19312) | > |13.601 | RTP (g711A) |RTP Num > packets:220 Duration:4.419s SSRC:0xA530C519 > | |(26056) ------------------> (19312) | > > After the 100 Trying is received the switch executes > switch_channel_perform_set_running_state (switch_channel.c) and the channel > call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is > generated. > > I would expect the channel call state to be set to RINGING only > after 180 Ringing is received from the far endpoint. > > Could anybody give me a hint what could be wrong or what steps to > take to figure out? I am already out of ideas. > > Thanks! > Hynek >
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
_________________________________________________________________________ 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
Good enough, thanks! Hynek On Sat, Feb 23, 2013 at 11:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Head is the most stable branch currently, its in final stage. We will not merge things until 1.2 is branched and 1.5 is tagged on head. On Feb 23, 2013 3:51 PM, "Hynek Cihlar" <hynek.cihlar@gmail.com> wrote:
With the HEAD it now works as expected, RINGING comes only after 180/183 is received from the originated endpoint.
Could you also merge the change into more stable branch? Do you want me to file a bug for that?
Hynek
On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I think I see why, nobody seems to use those callstate events enough to notice anything wrong with them, Try latest HEAD.
On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
Yes, it is outboubd from FS to somewhere else. Here's the console output http://pastebin.com/Awme44SZ.
Hynek
On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The data provided is not really relevant to FS. If you want a diagnosis of what is happening in FS you need to provide FS logs from the console.
I am not even 100% sure which direction this is but I think you mean outbound from FS to somewhere else.
console loglevel debug sofia global siptrace on
On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre@gmail.com>wrote:
There are some use-cases.
The main one would be that you might get FS to generate a ringback tone if you're using ignore_early_media=true since otherwise the caller would never hear any ringing.
-Steve
On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar@gmail.com>wrote:
> Is there even a valid use case for Freeswitch to set a channel to > RINGING before the actual ring is signaled by the far endpoint? What other > evidence would be helpful to diagnose the issue? > > Hynek > > > On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar < > hynek.cihlar@gmail.com> wrote: > >> When originating a call the respective call channel's call state is >> set to RINGING right after progress 100 is received. >> >> Here's the captured flow: >> >> |Time | <src ip> | >> | | | <dst ip> | >> >> |2.848 | INVITE SDP (g711A g711U GSM >> telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed >> To:<sip:endpoint removed >> | |(5080) ------------------> (5060) | >> |2.848 | 407 Proxy Authentication Required |SIP >> Status >> | |(5080) <------------------ (5060) | >> |2.849 | ACK | |SIP Request >> | |(5080) ------------------> (5060) | >> |2.849 | INVITE SDP (g711A g711U GSM >> telephone-eventRTP...e-101 CN) |SIP From: "" <sip:endpoint removed >> To:<sip:endpoint removed >> | |(5080) ------------------> (5060) | >> |2.850 | 100 Trying| |SIP Status >> | |(5080) <------------------ (5060) | >> |13.444 | 180 Ringing |SIP Status >> | |(5080) <------------------ (5060) | >> |13.445 | 183 Session Progress SDP (g711A g711U GSM >> tele...ne-eventRTPType-101) |SIP Status >> | |(5080) <------------------ (5060) | >> |13.445 | RTP (g711A) |RTP Num >> packets:230 Duration:4.574s SSRC:0x1E777E26 >> | |(26056) <------------------ (19312) | >> |13.601 | RTP (g711A) |RTP Num >> packets:220 Duration:4.419s SSRC:0xA530C519 >> | |(26056) ------------------> (19312) | >> >> After the 100 Trying is received the switch executes >> switch_channel_perform_set_running_state (switch_channel.c) and the channel >> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is >> generated. >> >> I would expect the channel call state to be set to RINGING only >> after 180 Ringing is received from the far endpoint. >> >> Could anybody give me a hint what could be wrong or what steps to >> take to figure out? I am already out of ideas. >> >> Thanks! >> Hynek >> > > > > _________________________________________________________________________ > 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 > >
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
participants (4)
-
Anthony Minessale
-
Gabriel Gunderson
-
Hynek Cihlar
-
Steven Ayre