Inbound call through Registered gateway
Hi all, I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call. I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call. I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in. Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense. public.xml <extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension> Gateway config <include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include> Thanks in advance for your help.
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile? On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
The INVITE is just dropped with a 408, I don't see any dialplan logging output. Your help is appreciated on this. SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
Enable the debug-level log ("/log 9" in fs_cli) and show us everything that happens when the INVITE is received. On 15 April 2014 22:34, Luis F Urrea <lfurrea@gmail.com> wrote:
The INVITE is just dropped with a 408, I don't see any dialplan logging output.
Your help is appreciated on this.
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
_________________________________________________________________________ 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
Well freeswitch@sipzag> console loglevel +OK console log level set to DEBUG However I only see output from the sofia siptrace on not regular freeswitch logs. Does that even make sense? On Tue, Apr 15, 2014 at 2:44 PM, Steven Ayre <steveayre@gmail.com> wrote:
Enable the debug-level log ("/log 9" in fs_cli) and show us everything that happens when the INVITE is received.
On 15 April 2014 22:34, Luis F Urrea <lfurrea@gmail.com> wrote:
The INVITE is just dropped with a 408, I don't see any dialplan logging output.
Your help is appreciated on this.
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
_________________________________________________________________________ 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
My bad, I was executing freeswitch through screen and it seems debug output is controled by fsctl debug_level 8 I do see output now but nothing like I have ever seen in FS freeswitch@sipzag> 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:334 Looking for zrtp-hash 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:289 Deciding whether to pass zrtp-hash between legs 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:291 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_ivr.c:1835 sofia/tone_detect/ xlite@129.sip.testy.com receive message [TRANSFER] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2693 sofia/tone_detect/xlite@129.sip.testy.com receive message [PROGRESS] 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:334 Looking for zrtp-hash 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:289 Deciding whether to pass zrtp-hash between legs 2014-04-15 23:41:43.338539 [ALERT] switch_core_media.c:294 No partner channel found, so not propagating zrtp-hash 2014-04-15 23:41:43.338539 [ALERT] sofia_media.c:92 sofia/tone_detect/ xlite@129.sip.testy.com receive message [RING_EVENT] 2014-04-15 23:41:43.338539 [ALERT] switch_channel.c:3430 sofia/tone_detect/ xlite@129.sip.testy.com receive message [AUDIO_SYNC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.338539 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.678664 [ALERT] switch_core_media.c:334 Looking for zrtp-hash 2014-04-15 23:41:43.678664 [ALERT] switch_core_media.c:289 Deciding whether to pass zrtp-hash between legs 2014-04-15 23:41:43.678664 [ALERT] switch_core_media.c:291 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.678664 [ALERT] switch_ivr.c:1835 sofia/tone_detect/ xlite@129.sip.testy.com receive message [TRANSFER] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.678664 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2839 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2854 sofia/tone_detect/xlite@129.sip.testy.com receive message [APPLICATION_EXEC_COMPLETE] 2014-04-15 23:41:43.698505 [ALERT] switch_core_session.c:2693 sofia/tone_detect/xlite@129.sip.testy.com receive message [PROGRESS] 2014-04-15 23:41:43.698505 [ALERT] switch_core_media.c:334 Looking for zrtp-hash 2014-04-15 23:41:43.698505 [ALERT] switch_core_media.c:289 Deciding whether to pass zrtp-hash between legs 2014-04-15 23:41:43.698505 [ALERT] switch_core_media.c:294 No partner channel found, so not propagating zrtp-hash On Tue, Apr 15, 2014 at 3:50 PM, Luis F Urrea <lfurrea@gmail.com> wrote:
Well
freeswitch@sipzag> console loglevel
+OK console log level set to DEBUG
However I only see output from the sofia siptrace on not regular freeswitch logs.
Does that even make sense?
On Tue, Apr 15, 2014 at 2:44 PM, Steven Ayre <steveayre@gmail.com> wrote:
Enable the debug-level log ("/log 9" in fs_cli) and show us everything that happens when the INVITE is received.
On 15 April 2014 22:34, Luis F Urrea <lfurrea@gmail.com> wrote:
The INVITE is just dropped with a 408, I don't see any dialplan logging output.
Your help is appreciated on this.
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
_________________________________________________________________________ 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
Enable the debug-level log ("/log 9" in fs_cli)
The command you ran affects mod_console which is the foreground console. fs_cli connects to the remote console via mod_event_socket, for it you set the logging level within your fs_cli client with its local /log command since different ESL clients can be using different logging levels. "/log 9" will show all logging messages On 15 April 2014 23:50, Luis F Urrea <lfurrea@gmail.com> wrote:
Well
freeswitch@sipzag> console loglevel
+OK console log level set to DEBUG
However I only see output from the sofia siptrace on not regular freeswitch logs.
Does that even make sense?
On Tue, Apr 15, 2014 at 2:44 PM, Steven Ayre <steveayre@gmail.com> wrote:
Enable the debug-level log ("/log 9" in fs_cli) and show us everything that happens when the INVITE is received.
On 15 April 2014 22:34, Luis F Urrea <lfurrea@gmail.com> wrote:
The INVITE is just dropped with a 408, I don't see any dialplan logging output.
Your help is appreciated on this.
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
_________________________________________________________________________ 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
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
Right what I was trying to say is that I am running this through the foreground console, and the output I pasted is exactly what is happening with that inbound call. On Tue, Apr 15, 2014 at 4:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
Enable the debug-level log ("/log 9" in fs_cli)
The command you ran affects mod_console which is the foreground console. fs_cli connects to the remote console via mod_event_socket, for it you set the logging level within your fs_cli client with its local /log command since different ESL clients can be using different logging levels.
"/log 9" will show all logging messages
On 15 April 2014 23:50, Luis F Urrea <lfurrea@gmail.com> wrote:
Well
freeswitch@sipzag> console loglevel
+OK console log level set to DEBUG
However I only see output from the sofia siptrace on not regular freeswitch logs.
Does that even make sense?
On Tue, Apr 15, 2014 at 2:44 PM, Steven Ayre <steveayre@gmail.com> wrote:
Enable the debug-level log ("/log 9" in fs_cli) and show us everything that happens when the INVITE is received.
On 15 April 2014 22:34, Luis F Urrea <lfurrea@gmail.com> wrote:
The INVITE is just dropped with a 408, I don't see any dialplan logging output.
Your help is appreciated on this.
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0 Via: SIP/2.0/UDP XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a From: "Sangoma Technologies" <sip:6502626901@129.sip.testy.com
;tag=g0FpDc3651FtN To: <sip:tone_detect@XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131 CSeq: 58462619 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "tone_detect" <sip:tone_detect@XX.XX.XX.XX ;party=calling;privacy=off;screen=no
On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
The point is that the gateway registers using a random port as the source port 52762 and then when the INVITE comes how would that be associated with the SIP profile?
On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea@gmail.com>wrote:
Hi all,
I am trying to receive a call through a gateway that is suing registration on external profile and I don't clearly understand how to configure the extension to receive the call.
I appears from the example.xml gateway configuration that the an extension name with the username used for gateway registration can be used to get the dialplan to accept the call.
I assume this needs to be configured on the public context, however I see the INVITE coming in, but the dialplan never kicks in.
Do I have to receive the INVITE at port 5080? If that is the case, registration doesn't really make sense.
public.xml
<extension name="tone_detect"> <condition field="destination_number" expression="^tone_detect$"> <action application="pre_answer"/> <action application="sleep" data="20000"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-goodbye.wav"/> <action application="hangup"/> </condition> </extension>
Gateway config
<include> <gateway name="tone_detect"> <param name="username" value="tone_detect"/> <...snip...> </include>
Thanks in advance for your help.
_________________________________________________________________________ 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
_________________________________________________________________________ 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 (2)
-
Luis F Urrea
-
Steven Ayre