outbound call, new comer
Hi, I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: <include> <gateway name="voicepulse"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="extension" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> ------------ and in the conf/dialplan/default.xml file I added: <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> ------------------ For inbound, I added <extension name="Voicepulse"> <!-- your provider or any name you'd like to call it --> <condition field="destination_number" expression="3334445555"> <!-- your DID for this gateway--> <action application="transfer" data="1001 XML default"/> </condition> </extension> ---------- And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... Would someone please give me some help to set this up. I am a bit confused with these. Thank you
In West Philadelphia born and raised... Voicepulse seems to be picky about number format. Trying doing full E.164 (+1). Also, make sure your realm is correct. What does a SIP debug look like? On 1/12/09, Will Smith <willbelair@yahoo.com> wrote:
Hi, I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes:
<include> <gateway name="voicepulse"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="extension" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include>
------------ and in the conf/dialplan/default.xml file I added:
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here --> <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension>
------------------
For inbound, I added
<extension name="Voicepulse"> <!-- your provider or any name you'd like to call it --> <condition field="destination_number" expression="3334445555"> <!-- your DID for this gateway--> <action application="transfer" data="1001 XML default"/> </condition> </extension>
----------
And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ...
Would someone please give me some help to set this up. I am a bit confused with these.
Thank you
_______________________________________________ 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
-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
Forgive me, I don't know how to turn on the SIP debug mode. This is what it say from FS command line: 2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000@192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000@192.168.2.104) Ended 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.2.104 [CS_HANGUP] --- On Mon, 1/12/09, Kristian Kielhofner <kristian.kielhofner@gmail.com> wrote: From: Kristian Kielhofner <kristian.kielhofner@gmail.com> Subject: Re: [Freeswitch-users] outbound call, new comer To: freeswitch-users@lists.freeswitch.org Date: Monday, January 12, 2009, 1:07 PM In West Philadelphia born and raised... Voicepulse seems to be picky about number format. Trying doing full E.164 (+1). Also, make sure your realm is correct. What does a SIP debug look like? On 1/12/09, Will Smith <willbelair@yahoo.com> wrote:
Hi, I am first time FS user, so it is a bit confused with all the setup. For
inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes:
<include> <gateway name="voicepulse"> <param name="username" value="3334445555"/> <param name="realm"
value="my_sip_provider.com"/>
<param name="password" value="3334445555"/> <param name="proxy"
value="my_sip_provider.com"/>
<param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="3334445555"/> <param name="realm"
value="my_sip_provider.com"/>
<param name="password" value="3334445555"/> <param name="extension" value="3334445555"/> <param name="proxy"
value="my_sip_provider.com"/>
<param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include>
------------ and in the conf/dialplan/default.xml file I added:
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444)
here -->
<extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension>
------------------
For inbound, I added
<extension name="Voicepulse"> <!-- your provider or any name you'd like to call it --> <condition field="destination_number" expression="3334445555"> <!-- your DID for this gateway--> <action application="transfer" data="1001 XML default"/> </condition> </extension>
----------
And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ...
Would someone please give me some help to set this up. I am a bit confused with these.
Thank you
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-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
TPORT_LOG=1 ./freeswitch /b On Jan 12, 2009, at 3:21 PM, Will Smith wrote:
Forgive me, I don't know how to turn on the SIP debug mode. This is what it say from FS command line:
2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000@192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000@192.168.2.104 ) Ended 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.2.104 [CS_HANGUP]
I always just use ngrep. As long as you aren't using TLS it works quite well: ngrep -d [device] -q -W byline SIP udp port 5060 Of course you can update your BPF syntax if you wish. On 1/12/09, Brian West <brian@freeswitch.org> wrote:
TPORT_LOG=1 ./freeswitch
/b
On Jan 12, 2009, at 3:21 PM, Will Smith wrote:
Forgive me, I don't know how to turn on the SIP debug mode. This is what it say from FS command line:
2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default 2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway 2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT 2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000@192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000@192.168.2.104) Ended 2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.2.104 [CS_HANGUP]
_______________________________________________ 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
-- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com
This is what I found in the log file. Also, if from command line I type: originate sofia/external/355@my_sip_provider.com 2009 , it will ring my extension and play some music (which is defined in the 2009.xml file) Thank you for all your help -------------------------------- 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [show_info] destination_number(18187188234) =~ /^9992$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [video_record] destination_number(18187188234) =~ /^9993$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [video_playback] destination_number(18187188234) =~ /^9994$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [delay_echo] destination_number(18187188234) =~ /^9995$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [echo] destination_number(18187188234) =~ /^9996$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [milliwatt] destination_number(18187188234) =~ /^9997$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [tone_stream] destination_number(18187188234) =~ /^9998$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [hold_music] destination_number(18187188234) =~ /^9999$/ 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex mismatch 2009-01-13 13:01:58 [DEBUG] mod_dialplan_xml.c:117 parse_exten() Regex: [qtioutbound] destination_number(18187188234) =~ /^(1{0,1}\d{10})$/ 2009-01-13 13:01:58 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/internal/1000@192.168.2.104) State Change CS_ROUTING -> CS_EXECUTE 2009-01-13 13:01:58 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/1000@192.168.2.104 [BREAK] ----------------------
participants (3)
-
Brian West
-
Kristian Kielhofner
-
Will Smith