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[Freeswitch-users] outbound call, new comer


 
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willbelair at yahoo.com
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PostPosted: Mon Jan 12, 2009 3:52 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

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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willbelair at yahoo.com
Guest





PostPosted: Mon Jan 12, 2009 4:23 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

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:
Quote:
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

Quote:
In West Philadelphia born and raised...Voicepulse seems to be picky about number format. Trying doing fullE.164 (+1). Also, make sure your realm is correct. What does a SIPdebug 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. Forinbound calls, I tried to add a voicepulse.xml in the sip_profiles/external withthe 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) youmay 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 orany name you'd like to call it -->> <condition field="destination_number"expression="3334445555"> <!-- your DID for this gateway-->> <action application="transfer" data="1001 XMLdefault"/>> </condition>> </extension>>> ---------->> And, if I dial 3334445555 from a softphone registered withmy_sip_provider, I got the message to the voice mail of 1001 - the 1001extension does not ring.> And if from 1001, I dial some real number like 18188892345, I got theerror: 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 confusedwith 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 Kielhofnerhttp://blog.krisk.orghttp://www.submityoursip.comhttp://www.astlinux.orghttp://www.star2star.com_______________________________________________Freeswitch-users mailing
listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
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brian at freeswitch.org
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PostPosted: Mon Jan 12, 2009 4:31 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

TPORT_LOG=1 ./freeswitch

/b

On Jan 12, 2009, at 3:21 PM, Will Smith wrote:
Quote:
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 ([email]sofia/internal/1000@192.168.2.104[/email]) [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 ([email]sofia/internal/1000@192.168.2.104[/email])) 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 ([email]sofia/internal/1000@192.168.2.104[/email]) [CS_HANGUP]
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kristian.kielhofner at...
Guest





PostPosted: Mon Jan 12, 2009 4:45 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

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:
Quote:


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

_______________________________________________
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
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kristian.kielhofner at...
Guest





PostPosted: Mon Jan 12, 2009 4:50 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

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:
Quote:

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

_______________________________________________
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
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willbelair at yahoo.com
Guest





PostPosted: Mon Jan 12, 2009 5:01 pm    Post subject: [Freeswitch-users] outbound call, new comer Reply with quote

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]


----------------------
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