Hi, Freeswitch-users. My dialplan: <extension name="Local_Extension"> <condition field="destination_number" expression="^([1-9]\d{4})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="bypass_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/${originate_disposition}.wav"/> </condition> </extension> If hangup cause is SUBSCRIBER_ABSENT or USER_NOT_REGISTERED, then playback don't work. Piece of log 2010-05-26 17:35:08.218253 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [INFO] mod_dptools.c:2356 Originate Failed. Cause: SUBSCRIBER_ABSENT EXECUTE sofia/internal/39663@10.110.1.8:5060 pre_answer() 2010-05-26 17:35:08.219282 [INFO] mod_dptools.c:746 Sending early media 2010-05-26 17:35:08.219282 [NOTICE] mod_sofia.c:2032 Pre-Answer sofia/internal/39663@10.110.1.8:5060! 2010-05-26 17:35:08.219282 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] EXECUTE sofia/internal/39663@10.110.1.8:5060 sleep(1000) 2010-05-26 17:35:08.219282 [DEBUG] sofia.c:4195 Channel sofia/internal/39663@10.110.1.8:5060 skipping state [early][183] 2010-05-26 17:35:09.219252 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] 2010-05-26 17:35:39.226271 [ERR] switch_core_io.c:121 sofia/internal/39663@10.110.1.8:5060 has no read codec. 2010-05-26 17:35:39.226271 [DEBUG] switch_core_session.c:1759 Application playback Requires media on channel sofia/internal/39663@10.110.1.8:5060! EXECUTE sofia/internal/39663@10.110.1.8:5060 playback(elight/SUBSCRIBER_ABSENT.wav) 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:185 sofia/internal/39663@10.110.1.8:5060 has executed the last dialplan instruction, hanging up. 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/39663@10.110.1.8:5060 [CS_EXECUTE] [NORMAL_CLEARING] What am I doing wrong? If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation. Thanks. -- Regard Sergey Scheglov
Hello, Brian. You wrote 27 may 2010, 2:52:05:
What rev are you running?
/b
_______________________________________________ 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's for reply, Brian [root@freeswitch]# fs_cli -x "version" FreeSWITCH Version 1.0.head (git-) :( I don't known why/ But "git clone" from 05/20/2010. -- Regard Sergey Scheglov
Hi, Brian. You wrote 27 may 2010 г., 2:52:05:
What rev are you running?
/b
On May 26, 2010, at 5:49 AM, Sergey Scheglov wrote:
Thank's for reply. I'm update git and FS. Now, [root@freeswitch]# fs_cli -x "version" FreeSWITCH Version 1.0.head (git-095815f 2010-05-26 20-58-16 -0500) The problem still remains. Regard's Sergey Scheglov
we can't tell why because you omitted the part of the log that may show why. also, why do you set continue on fail twice in a row? On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid@eltc.ru> wrote:
Hi, Freeswitch-users.
My dialplan: <extension name="Local_Extension"> <condition field="destination_number" expression="^([1-9]\d{4})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="bypass_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@$ {domain_name}"/> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/$ {originate_disposition}.wav"/> </condition> </extension>
If hangup cause is SUBSCRIBER_ABSENT or USER_NOT_REGISTERED, then playback don't work.
Piece of log
2010-05-26 17:35:08.218253 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2010-05-26 17:35:08.218253 [INFO] mod_dptools.c:2356 Originate Failed. Cause: SUBSCRIBER_ABSENT EXECUTE sofia/internal/39663@10.110.1.8:5060 pre_answer() 2010-05-26 17:35:08.219282 [INFO] mod_dptools.c:746 Sending early media 2010-05-26 17:35:08.219282 [NOTICE] mod_sofia.c:2032 Pre-Answer sofia/internal/39663@10.110.1.8:5060! 2010-05-26 17:35:08.219282 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] EXECUTE sofia/internal/39663@10.110.1.8:5060 sleep(1000) 2010-05-26 17:35:08.219282 [DEBUG] sofia.c:4195 Channel sofia/ internal/39663@10.110.1.8:5060 skipping state [early][183] 2010-05-26 17:35:09.219252 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663@10.110.1.8:5060 [BREAK] 2010-05-26 17:35:39.226271 [ERR] switch_core_io.c:121 sofia/internal/ 39663@10.110.1.8:5060 has no read codec. 2010-05-26 17:35:39.226271 [DEBUG] switch_core_session.c:1759 Application playback Requires media on channel sofia/internal/ 39663@10.110.1.8:5060! EXECUTE sofia/internal/39663@10.110.1.8:5060 playback(elight/ SUBSCRIBER_ABSENT.wav) 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:185 sofia/internal/39663@10.110.1.8:5060 has executed the last dialplan instruction, hanging up. 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/39663@10.110.1.8:5060 [CS_EXECUTE] [NORMAL_CLEARING]
What am I doing wrong?
If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation.
Thanks. -- Regard Sergey Scheglov
_______________________________________________ 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
Hi, Michael. You wrote 27 may 2010 г., 12:30:25:
we can't tell why because you omitted the part of the log that may show why. also, why do you set continue on fail twice in a row?
On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid@eltc.ru> wrote: Thanks for reply,
"continue_on_fail=true" I removed (my fault) and altered bypass_media (my reasons) Now dialplan is <extension name="Local_Extension"> <condition field="destination_number" expression="^([1-9]\d{4})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="bypass_media=${cond(${profile_index} == 1 ? true : false)}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/${originate_disposition}.wav"/> </condition> </extension> Full call log here http://pastebin.freeswitch.org/13059 Also, in internal profile exist: inbound-late-negotiation true disable-transcoding true and I used xml_curl (for directory). -- С уважением, Сергей Щеглов ISP Good Line e-mail: sid@eltc.ru icq: 295481909 тел. +79234872136
playback executes on line 87 of your trace. If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or the phone you are on does not support early media, change your pre-answer before the playback to answer to verify. Try adding sofia profile internal siptrace on to see the sip traffic too. 2010/5/27 Sergey Scheglov <sid@eltc.ru>
Hi, Michael.
You wrote 27 may 2010 г., 12:30:25:
we can't tell why because you omitted the part of the log that may show why. also, why do you set continue on fail twice in a row?
On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid@eltc.ru> wrote: Thanks for reply,
"continue_on_fail=true" I removed (my fault) and altered bypass_media (my reasons) Now dialplan is <extension name="Local_Extension"> <condition field="destination_number" expression="^([1-9]\d{4})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="bypass_media=${cond(${profile_index} == 1 ? true : false)}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/${originate_disposition}.wav"/> </condition> </extension>
Full call log here http://pastebin.freeswitch.org/13059 Also, in internal profile exist: inbound-late-negotiation true disable-transcoding true and I used xml_curl (for directory). -- С уважением, Сергей Щеглов ISP Good Line e-mail: sid@eltc.ru icq: 295481909 тел. +79234872136
_______________________________________________ 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 <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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite. instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works. 2010/5/27 Sergey Scheglov <sid@eltc.ru>
Hi, Anthony.
You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.
Try adding sofia profile internal siptrace on to see the sip traffic too.
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or
the phone you are on does not support early media,
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is:
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set).
change your pre-answer before the playback to answer to verify.
If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065
Thanks again :)
--
Regard's
Sergey Scheglov
_______________________________________________ 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 <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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hi, Anthony. Thanks, thanks, thanks )) You wrote 28 may 2010., 1:46:05:
it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
Do I understand correctly that the dialplan is not correct?
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
As I wrote in first message: "If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation." And two phones that support codecs, which not present in internal profile, not be able to call each other. Best regards Sergey Scheglov
not sure what you mean but i'm simply explaining to you that you cannot re-invite to change the media in the SOA before the initial call was setup. On Thu, May 27, 2010 at 2:12 PM, Sergey Scheglov <sid@eltc.ru> wrote:
Hi, Anthony.
Thanks, thanks, thanks )) You wrote 28 may 2010., 1:46:05:
it's against the SIP spec to re-negotiate media before you have answered. the transaction to negotiate the SDP from the original call has not completed so it's illegal to send a re-invite.
Do I understand correctly that the dialplan is not correct?
instead you should use bypass_media_after_bridge=true so the bypass only happens when the bridge works.
As I wrote in first message: "If set bypass_media_after_bridge instead bypass_media, then works fine, BUT changing codec negotiation." And two phones that support codecs, which not present in internal profile, not be able to call each other.
Best regards Sergey Scheglov
_______________________________________________ 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 <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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hi, Anthony. Thank's for reply. You wrote 27 may 2010 г., 20:34:15:
playback executes on line 87 of your trace.
yes, but executed by log after 30 seconds after line 84 and immediately hangup. Note, duration wav file - 4 sec.
If you do not hear the audio, It means the re-establishment of media fails somehow based on your topology or the phone you are on does not support early media, change your pre-answer before the playback to answer to verify.
I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case. If dialplans is: <extension name="fail_balance"> <condition field="${blabla}" expression="^\-1$"> <action application="pre_answer"/> <action application="sleep" data="1000"/> <action application="playback" data="elight/neg_balance.wav"/> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/> </condition> </extension> then work's fine, no problems (bypass_media=true not set). If set answer, problem disappears. But answer it's not for my case.
Try adding sofia profile internal siptrace on to see the sip traffic too.
Call log with sip trace http://pastebin.freeswitch.org/13065 Thanks again :) Best regard's Sergey Scheglov
participants (4)
-
Anthony Minessale
-
Brian West
-
Michael Jerris
-
Sergey Scheglov