July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon in 3 WEEKS
by Anthony Minessale 15 Jul '08

15 Jul '08

I wanted to let everyone know that ClueCon 2008 is just under 3 weeks away! Everyone who is planning to attend: Please make sure you are registered and have your rooms booked. We have an overflowing schedule of speakers but we could still probably squeeze in a few more lightening talks if anyone wants to speak. This year is shaping up to be the best yet with developers or reps from most of the popular Open Source VoIP applications represented including: FreeSWITCH(tm) (naturally) ;) OpenSER OPAL sipX/SIPfoundry Asterisk Yate Bayonne SofaSWITCH We also have reps or developers from several hardware vendors including: Sangoma PIKA Digium Dialogic SNOM OCTASIC We also have 2 professional voice firms attending including: GM Voices (creator of the famous Cepstral Callie and all the FreeSWITCH(tm) prompts) Allison Smith (The voice of Asterisk(tm)) -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Playing an audio file to end destination before bridging the call
by Adnan Barakat 15 Jul '08

15 Jul '08

Hi All, I'm looking for a way to play an audio file to the end destination before bridging the call, eg. below is an example in Asterisk which I'm trying to replicate in FS. exten => _0ZXXXXXXXXX,1,Dial(SIP/outsip/${dest},30,rA(/srv/files/whisperfile)) Is this possible in FS? Any feedback would be highly appreciated. Adnan

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send_dtmf problems
by John Wehle 15 Jul '08

15 Jul '08

> Try a sleep after the answer. Will do ... just curious as to why and for how long? >> 1) When I dial the extension from a Grandstream GXP-2000 ... >> How do I configure FreeSWITCH to send both RTP digits and inband audio? > The VoIP phone has no reason to reproduce any DTMF. I understand that the VoIP phone has no reason to create DTMF tones from the RTP digits sent by FreeSWITCH. What I was asking is how to you configure FreeSWITCH to generate inband DTMF tones in addition to the RTP digits. I.e. this particular phone can be configured to send DTMF digits to FreeSWITCH as any combination of inband audio, RTP digits, and / or SIP info. Going in the opposite direction I see where in the FreeSWITCH sofia xml configuration type you can set the dtmf-type as either rfc2833 or as SIP info however I don't see an option for using both nor do I see an option for generating inband DTMF audio. Note: I was just using send_dtmf with the VoIP phone for testing purposes ... this application doesn't actually need to send DTMF to a VoIP phone. >> 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, > Is this using OpenZAP? Yes. In my particular application the System 25 PBX connects to FreeSWITCH using OpenZAP running on a Sangoma A204DX card. At the end of a call sent to voicemail I need FreeSWITCH to send some DTMF tones to the PBX in order to turn on / off the message waiting indicator. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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outbound fxo line pooling
by John Wehle 15 Jul '08

15 Jul '08

> a or A depending on if you want top down.. or bottom up. Okay, so with OpenZAP I can use either a number to refer to a specific port or A to refer to any available port in a span. Other than using a number or A (and a) OpenZAP currently has no mechanism for defining explicit pools / port ranges. So to do something like have a four port card setup with inbound calls using all four ports and outbound calls using only ports 2 - 4 would I need to configure the card as two spans with span 1 containing port 1, and span 2 containing ports 2 - 4? Then I route outbound calls to span 2 using the A port (aka any available port on span 2)? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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SIPfoundry cooperates with the FreeSWITCH project
by Brian West 15 Jul '08

15 Jul '08

Check it out!! http://digg.com/software/SIPfoundry_cooperates_with_the_FreeSWITCH_project Check it out... Digg it and pass it around to 10 friends. ;) /b Brian West sip:brian@freeswitch.org

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Call-Blocking application
by Alois Komenda 15 Jul '08

15 Jul '08

Hello, how can I realize a Call-Blocking application with FreeSWITCH? The application should look up the caller in a kind of blacklist or delegate the lookup to another application. Every incoming call has to be blocked if the caller is on the blacklist. Is it possible to realize this as a application which is called from the dialplan? As the dialplan is parsed only once for a call, it won't work if the application is called from an action tag, I guess. The outcome of the application can't influence any following action, right? What about calling the application in the condition tag? Would that count as an "abuse" as mentioned in the wiki? What method can you recommend to block incoming calls depending on the originator? Regards, -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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outbound fxo line pooling
by John Wehle 15 Jul '08

15 Jul '08

I currently have: <extension name="outgoing-fxo"> <condition field="destination_number" expression="^(4[0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> <action application="bridge" data="openzap/1/1/${dialed_ext}"/> </condition> </extension> which routes calls to extension 4XX out the first openzap line. How do I set up a pool of openzap lines and route calls to extension 4XX to any available openzap line? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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send_dtmf problems
by John Wehle 15 Jul '08

15 Jul '08

I need to send some dtmf tones after handling vmail for the System 25 so I've set up (for testing purposes) an extension which should just answer the phone and play dtmf: <extension name="system25_vmail"> <condition field="destination_number" expression="^(5590)|(55[1-8])$"> <action application="answer"/> <action application="send_dtmf" data="1234567890@500"/> <action application="sleep" data="5000"/>  </condition> </extension> 1) When I dial the extension from a Grandstream GXP-2000 a network trace shows the digits being sent from FreeSWITCH as RTP events, however no dtmf audio tones are sent. How do I configure FreeSWITCH to send both RTP digits and inband audio? 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, after five seconds it sends two tones, and then disconnects. How do I configure FreeSWITCH to send all the dtmf tones? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Problem: call routing via external gateway not working
by Birgit Arkesteijn 14 Jul '08

14 Jul '08

Hi, I'm running FreeSWITCH Version 1.0.trunk (498:8901) on Suse 10.0 (x86_64). I'm a novice in setting up the configuration, so sorry if this is a trivial one. I'm trying to get the following test setup to work: 1. SIP desktop phone 2. -> Asterisk on risk.westhawk.co.uk 3. -> FreeSWITCH on apso.westhawk.co.uk (using xml_curl to get a dialplan) 4. -> Asterisk on risk.westhawk.co.uk 5. -> another SIP desktop phone I'm fine up to (and including) step 3: - I can see the incoming call - I can see the dialplan being accessed - I can see the dialplan being matched - I can see from the freeswitch log it's trying to bridge I'm stuck going to step 4: - I cannot see any SIP packets going out back to risk We checked our firewall log files, but no packets are dropped to risk (as far as we can tell). What I've configured is: - a user in directory/ so the incoming call works - an external gateway, called 'doneright', for the outgoing leg Below are - snippets from the freeswitch.log file, - some 'sofia status' commands on the console, - the content of my doneright.xml external gateway file The 'doneright' gateway doesn't currently register at risk, because we don't think we need to. When we omitted the 'register' setting by accident in the doneright.xml file, we noticed sip registration requests coming in at risk. I'm very puzzled by this all. On a different machine, we've done SIP routing to an external (then called 'outbound') SIP provider without a problem. Does anyone have a clue or some pointers how I can approach this problem? Thanks, Birgit *** Freeswitch log file: * (incoming call): 2008-07-09 11:36:00 [NOTICE] sofia.c:1514 config_sofia() Adding Alias [apso.westhawk.co.uk] for profile [internal] 2008-07-09 11:46:24 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/apso(a)192.67.4.83[466eaeee-4da4-11dd-b14f-7bc0d4e408b2] 2008-07-09 11:46:24 [DEBUG] sofia.c:1748 sofia_handle_sip_i_state() Channel sofia/internal/apso(a)192.67.4.83 entering state [received] * (dialplan): 2008-07-09 11:35:34 [NOTICE] mod_xml_curl.c:300 do_config() Binding [doneright-voipuri] XML Fetch Function [http://apso.westhawk.co.uk/test_dialplan/static_dialplan.xml][dialplan] 2008-07-09 11:46:24 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Birgit Arkesteijn->801@default 2008-07-09 11:46:24 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(801) =~ /^801$/ * (bridge): 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/internal/apso(a)192.67.4.83 Execute bridge(sofia/gateway/doneright/1234) * (external gateway): 2008-07-09 11:46:24 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/1234 [467a1040-4da4-11dd-b14f-7bc0d4e408b2] 2008-07-09 11:46:24 [DEBUG] mod_sofia.c:1870 sofia_outgoing_channel() sofia/external/1234 State Change CS_NEW -> CS_INIT 2008-07-09 11:46:24 [DEBUG] switch_core_session.c:720 switch_core_session_signal_state_change() Kill sofia/external/1234 [BREAK] 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/1234 Running State Change CS_INIT 2008-07-09 11:46:24 [DEBUG] switch_core_state_machine.c:415 switch_core_session_run() (sofia/external/1234) State INIT 2008-07-09 11:46:24 [DEBUG] mod_sofia.c:80 sofia_on_init() sofia/external/1234 SOFIA INIT 2008-07-09 11:46:44 [DEBUG] sofia.c:194 sofia_event_callback() event [nua_i_state] status [487][Request Terminated] session: sofia/internal/apso(a)192.67.4.83 *** Console: > sofia status API CALL [sofia(status)] output: Name Type Data State ========================================================================== external profile sip:mod_sofia@192.67.4.56:5080 RUNNING (0) internal profile sip:mod_sofia@192.67.4.56:5060 RUNNING (0) nat profile sip:mod_sofia@192.67.4.56:5070 RUNNING (0) doneright gateway sip:apso@risk.westhawk.co.uk NOREG default alias internal ALIASED apso.westhawk.co.uk alias internal ALIASED outbound alias external ALIASED ========================================================================== 3 profiles 3 aliases > sofia status profile external API CALL [sofia(status profile external)] output: ========================================================================== Name external Domain Name external DBName sofia_reg_external Dialplan XML RTP-IP 192.67.4.56 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.67.4.56 Ext-SIP-IP 192.67.4.56 URL sip:mod_sofia@192.67.4.56:5080 BIND-URL sip:mod_sofia@192.67.4.56:5080;maddr=192.67.4.56 HOLD-MUSIC local_stream://moh CODECS PCMU,PCMA,GSM TEL-EVENT 101 CNG 13 SESSION-TO 0 MAX-DIALOG 0 Registrations: ========================================================================== ========================================================================== > sofia status gateway doneright API CALL [sofia(status gateway doneright)] output: ========================================================================== Name doneright Scheme Digest Realm risk.westhawk.co.uk Username apso Password yes From <sip:apso@risk.westhawk.co.uk;transport=udp> Contact <sip:apso@192.67.4.56:5080;transport=udp> To sip:apso@risk.westhawk.co.uk Proxy sip:risk.westhawk.co.uk Context default Expires 600 Freq 598 Ping 0 PingFreq 0 State NOREG Status UP ========================================================================== *** /usr/local/freeswitch/conf/sip_profiles/external/doneright.xml: <include> <gateway name="doneright"> <param name="username" value="apso"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="risk.westhawk.co.uk"/> <param name="proxy" value="risk.westhawk.co.uk"/> <param name="register" value="false"/> <param name="expire-seconds" value="600"/> </gateway> </include> -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Freeswitch on OpenBSD
by Mucker 14 Jul '08

14 Jul '08

Hi All, Anybody else trying to get Freeswitch up and running on OpenBSD? Following the install instructions for FreeBSD "seems" to work. But, when I start freeswitch I see a critical error: 2008-07-14 16:56:02 [CRIT] switch_loadable_module.c:751 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **File not found** # ls -al /usr/local/freeswitch/mod/ | grep sofia -rw-r--r-- 1 root wheel 7907294 Jul 14 11:02 mod_sofia.a -rw-r--r-- 1 root wheel 1142 Jul 14 11:02 mod_sofia.la #

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