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missing Diversion header in 300 and 302 messages
by Tihomir Culjaga 04 Jun '11

04 Jun '11

hello, i found that FS doesn't transparently map Diversion header from INVITE to 300|302 message in a case of diversion. i have changed mod_sofia.c to adjust this: char *added_headers = sofia_glue_get_extra_headers(channel, SOFIA_SIP_HEADER_PREFIX); --- 2026,2037 ---- if (argc > 1) { nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES, SIPTAG_CONTACT_STR(dest), ! TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)), ! TAG_IF(!zstr(added_headers), SIPTAG_HEADER_STR(added_headers)), TAG_END()); } else { nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY, SIPTAG_CONTACT_STR(dest), ! TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)), ! TAG_IF(!zstr(added_headers), SIPTAG_HEADER_STR(added_headers)), TAG_END()); } Now 300|302 message contains the original header that was present in INVITE. since i didn't get any response whether this is a "feature" or a bug ... i went into the code and changed it myself. Please check the patch attached and see if it is possible/feasible/acceptable merge it into FS trunk. Regards, Tihomir. BTW: macros are cool while bringing headache in the same time :=)

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missing Diver sion header in 300, 302 message
by Tihomir Culjaga 04 Jun '11

04 Jun '11

hello, i found that FS doesn't transparently map Diversion header from INVITE to 300|302 message in a case of diversion. i have changed mod_sofia.c to adjust this: char *added_headers = sofia_glue_get_extra_headers(channel, SOFIA_SIP_HEADER_PREFIX); --- 2026,2037 ---- if (argc > 1) { nua_respond(tech_pvt->nh, SIP_300_MULTIPLE_CHOICES, SIPTAG_CONTACT_STR(dest), ! TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)), ! TAG_IF(!zstr(added_headers), SIPTAG_HEADER_STR(added_headers)), TAG_END()); } else { nua_respond(tech_pvt->nh, SIP_302_MOVED_TEMPORARILY, SIPTAG_CONTACT_STR(dest), ! TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)), ! TAG_IF(!zstr(added_headers), SIPTAG_HEADER_STR(added_headers)), TAG_END()); } Now 300|302 message contains the original header that was present in INVITE. since i didn't get any response whether this is a "feature" or a bug ... i went into the code and changed it myself. Please check the patch attached and see if it is possible/feasible/acceptable merge it into FS trunk. Regards, Tihomir. BTW: macros are cool while bringing headache in the same time :=)

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FreeSWITCH Conference Call Today
by Michael Collins 01 Jun '11

01 Jun '11

Hello all! Here is today's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_01 We have a guest from Shacast.com (Vuong Nguyen) who will be talking to us. Visit http://www.shacast.com and click on the PowerPoint download link to download the PPT that Vuong will be referencing for his presentation. Thanks, Michael

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ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

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Call For Help: FreeSWITCH Cookbook
by Michael Collins 31 May '11

31 May '11

Hello FreeSWITCHers! We have a need for some assistance with a few of our recipes for the cookbook. If you are able to write technical documentation in English and have first-hand knowledge of these topics then you might be in a position to help: CDRs: Parsing XML CDRs Handling A and B leg CDRs Using a web server to handle XML CDRs Event Socket: Inbound ESL connections (basic how-to) Outbound ESL connections (basic how-to) Launch an outbound call with inbound event socket & ESL Handle inbound call with socket app & ESL Reacting to events (events for "my call" vs. system-wide events, etc.) Misc: Presence for BLF and SLA Presence for FIFO agent status If you think you've got the chops to write one or more recipes for the cookbook then email me off list and we'll discuss the specifics. I will get you all the information you need to get started. Thanks! -Michael

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Re: [Freeswitch-dev] VXML parser
by Avni Mahajan 28 May '11

28 May '11

> > Hi, > > I am looking for a vxml (voicexml) parser in python language. Need to use > the parsed vxml tags and interact with freeswitch to run IVR. Can anyone > help me with any kind of opensource vxml parsers? > > Regards, > Avni Mahajan

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Accessing rtp_sessions (switch_rtp) from modules
by Gustav Koller 26 May '11

26 May '11

Hello, I'm new to FreeSwitch (developement). I'm currently writing a FreeSwitch module where I would like to access the zrtp_stream property of rtp streams (located in the switch_rtp struct/switch_rtp.c) from a module. Example: switch_rtp_t *rtp_session; rtp_session = switch_channel_get_private(channel, "__zrtp_audio_rtp_session"); zrtp_stream_t *zrtp_stream; zrtp_stream = rtp_session->zrtp_stream; The last line fails with "error: dereferencing pointer to incomplete type" It something I expected. The struct is defined in switch_rtp.c, which can't be included. So I can access the properties only from withing switch_rtp.c. Am I missing something? Is there another way to access the active zrtp_streams ? Best Regards, Gustav Koller

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extra sdp is always added in sofia_answer_channel ?
by Matteo 26 May '11

26 May '11

Hi, don't know if is a bug/typo or is correct, so I'm posting here. I'm noticed that in sofia_answer_channel (@mod_sofia.c:738, latest git) NUTAG_INCLUDE_EXTRA_SDP(1) is always added. Since we're into soa mode, maybe it should be added only by testing if 100rel is disabled, so it should be: --- mod_sofia.c.orig 2011-05-26 14:16:34.000000000 +0200 +++ mod_sofia.c 2011-05-26 14:16:47.000000000 +0200 @@ -734,7 +734,8 @@ SIPTAG_CONTACT_STR(tech_pvt->reply_contact), SIPTAG_CALL_INFO_STR(switch_channel_get_variable(tech_pvt->channel, SOFIA_SIP_HEADER_PREFIX "call_info")), SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str), - SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),NUTAG_INCLUDE_EXTRA_SDP(1), + SOATAG_REUSE_REJECTED(1), SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"), + TAG_IF(sofia_test_pflag(tech_pvt->profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)), TAG_IF(!zstr(extra_headers), SIPTAG_HEADER_STR(extra_headers)), TAG_IF(switch_stristr("update_display", tech_pvt->x_freeswitch_support_remote), SIPTAG_HEADER_STR("X-FS-Support: " FREESWITCH_SUPPORT)), TAG_END()); Or is correct to add it always? I've had to fix that because my upstream provider requires 100rel and does not want sdp in 200OK if is not changed from last progress message. Regards, Matteo

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Late response from Erlang to dialplan request causing Freeswitch to shut down
by Mark Sobkow 26 May '11

26 May '11

Platform: Ubuntu 10.04 amd64, 8GB RAM, 500GB HDD, quad-core CPU Configured to use SIP trunking as a slave off an Asterisk server with a T1 connection to the outside world We're using Erlang and PostgreSQL to provide responses to Freeswitch configuration requests and to manage our call queues, operator logins, etc. With one operator, the system runs fine, so we know the basic coding we've done works. However, with only a couple or three operators on the line and concurrent dialing going on, we're getting a consistent problem where the responses to requests for dialplan configurations are getting seriously delayed -- by well over a minute (an eternity in software terms.) I had suspected that there may be some sort of blocking IO going on when we issue the originate command (which doesn't return control to Erlang for up to 30 seconds), but I'm seeing other Freeswitch requests for directory information getting handled during this same time period, so the communications between Freeswitch and Erlang do support concurrent requests. The only thing I've been able to find in the mail list logs or wiki that seem relevant is a comment someone made today about fixing the TCP transmissions in Freeswitch so that they'll retry the transmission if the IP stream is clogged and would result in a blocking call. (i.e. FS used to be unreliable about sending out messages before this patch was done.) It's worth noting that Erlang threads are quite busy during the minute between FS issueing the request and Erlang responding to it. However, the thread/Erlang process that handles the configuration requests is idle, so it _should_ be able to respond in a much timelier fashion. It looks to me like there is some sort of delay between FS issuing the request and Erlang getting the message into it's processing queue. Any suggestions/ideas as to what might be causing this? And more importantly -- what can we do to fix or workaround the issue? And no, I have not downloaded and installed the latest version of Freeswitch from git. I'm trying to avoid that if possible because it'll cost me two weeks of regression testing and I need to have this out to a beta site by the 31st, which doesn't allow enough time to do the regression tests if we do an upgrade (and only a fool ships untested software to paying customers.) If someone can confirm that there was a fix checked into git for this issue or a related one, then I can justify the need for an upgrade and regression testing to my boss (though he won't be happy about the delay.) Here's the relevent snippet of the fs_cli log: 2011-05-26 11:45:50.648334 [INFO] mod_dialplan_xml.c:331 Processing Gotham <5311>->219 in context public 2011-05-26 11:45:50.648334 [ERR] mod_dialplan_xml.c:353 Open of dialplan failed 2011-05-26 11:45:50.648334 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-05-26 11:45:50.648334 [DEBUG] switch_channel.c:2535 (sofia/external/5311@10.77.0.12:6080) Callstate Change RINGING -> HANGUP 2011-05-26 11:45:50.657061 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/external/5311@10.77.0.12:6080 [CS_ROUTING] [NO_ROUTE_DESTINATION] And here's the log from where our response is finally being issued to the request, long after FS has decided to shut down because it's not getting responses from the configuration services: 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Invoking xml_fetch() 2011-05-26 11:47:45 DEBUG <0.28104.15>: pbx_bind_cfg:xml_fetch( {fetch, dialplan, Tag=undefined, Key=undefined, Value=undefined, Params=...} ) 2011-05-26 11:47:46 DEBUG <0.28104.15>: pbx_bind_cfg:fetch() Response XML is <document type="freeswitch/xml"> <section name="dialplan" description="Dial Plan For FreeSwitch"> <context name="default"> <extension required_name="ConferenceDialer"> <condition field="destination_number" expression="^8(\d\d\d)$"> <action application="conference" data="conf$0@dialer"/> </condition> </extension> <extension required_name="dialer_park"> <condition field="destination_number" expression="^park$"> <action application="fifo" data="park@${domain_name} in" /> </condition> </extension> <extension required_name="park"> <condition field="destination_number" expression="^7000$"> <action application="set" data="fifo_music=$${hold_music}" /> <action application="fifo" data="7000@${domain_name} in" /> </condition> </extension> <extension required_name="unpark"> <condition field="destination_number" expression="^7001$"> <action application="answer" /> <action application="fifo" data="7000@${domain_name} out nowait" /> </condition> </extension> <extension required_name="InternalFS"> <condition field="destination_number" expression="^(\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0%pursuit.marketel"/> </condition> </extension> <extension required_name="RatsExtension"> <condition field="destination_number" expression="^(\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> <extension required_name="LocalCall"> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> <extension required_name="LongDistanceCall"> <condition field="destination_number" expression="^91(\d\d\d\d\d\d\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> </context> <context name="public"> <extension required_name="Operator"> <condition field="destination_number" expression="^(0)$"> <action application="bridge" data="sofia/external/160(a)rats.marketel"/> </condition> </extension> <extension required_name="OperatorLogin"> <condition field="destination_number" expression="^620(\d)$"> <action application="answer" /> <action application="set" data="chain_id=$1" /> <action application="set" data="hangup_after_bridge=false" /> <action application="play_and_get_digits" data="4 8 1 30000 # /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav operator_pin \d+" /> <action application="log" data="INFO operator_pin is ${operator_pin}, chain_id is ${chain_id}" /> <action application="erlang" data="pbx_op_logged_in pursuit@pursuit" /> </condition> </extension> <extension required_name="OperatorLogin2"> <condition field="destination_number" expression="^62(\d\d)$"> <action application="answer" /> <action application="set" data="chain_id=$1" /> <action application="set" data="hangup_after_bridge=false" /> <action application="play_and_get_digits" data="4 8 1 30000 # /usr/lib/pursuit/lib/pbx-1.2/priv/welcome_please_enter_pin-8000.wav /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_pin_try_again-8000.wav operator_pin \d+" /> <action application="log" data="INFO operator_pin is ${operator_pin}, chain_id is ${chain_id}" /> <action application="erlang" data="pbx_op_logged_in pursuit@pursuit" /> </condition> </extension> <extension required_name="CollectCallResult"> <condition field="destination_number" expression="^63(\d\d)$"> <action application="answer" /> <action application="set" data="hangup_after_bridge=false" /> <action application="play_and_get_digits" data="1 2 1 30000 # /usr/lib/pursuit/lib/pbx-1.2/priv/please_enter_result_code-8000.wav /usr/lib/pursuit/lib/pbx-1.2/priv/invalid_result_code-8000.wav call_result \d+" /> <action application="log" data="INFO call_result is ${call_result}" /> <action application="erlang" data="pbx_call_result pursuit@pursuit" /> </condition> </extension> <extension required_name="DummyCallResult"> <condition field="destination_number" expression="^6400$"> <action application="answer" /> <action application="erlang" data="pbx_dummy_call_result pursuit@pursuit" /> </condition> </extension> <extension required_name="CustomerCallin"> <condition field="destination_number" expression="^650(\d)$"> <action application="answer" /> <action application="set" data="chain_id=$1" /> <action application="erlang" data="pbx_cust_called pursuit@pursuit" /> </condition> </extension> <extension required_name="CustomerCallin2"> <condition field="destination_number" expression="^65(\d\d)$"> <action application="answer" /> <action application="set" data="chain_id=$1" /> <action application="erlang" data="pbx_cust_called pursuit@pursuit" /> </condition> </extension> <extension required_name="TestIVR"> <condition field="destination_number" expression="^6600$"> <action application="answer" /> <action application="ivr" data="demo_ivr" /> </condition> </extension> <extension required_name="eavesdrop"> <condition field="destination_number" expression="^67(\d\d\d\d)$"> <action application="answer"/> <action application="set" data="eavesdrop_extension=$1" /> <action application="erlang" data="pbx_eavesdrop pursuit@pursuit" /> </condition> </extension> <extension required_name="park"> <condition field="destination_number" expression="^6003$"> <action application="fifo" data="6003@${domain_name} in" /> </condition> </extension> <extension required_name="unpark"> <condition field="destination_number" expression="^6004$"> <action application="answer" /> <action application="fifo" data="6003@${domain_name} out wait" /> </condition> </extension> <extension required_name="ConferenceDialer"> <condition field="destination_number" expression="^8(\d\d\d)$"> <action application="conference" data="conf$0@dialer"/> </condition> </extension> <extension required_name="InternalFS"> <condition field="destination_number" expression="^(\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0%10.77.0.12"/> </condition> </extension> <extension required_name="RatsExtension"> <condition field="destination_number" expression="^(\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> <extension required_name="LocalCall"> <condition field="destination_number" expression="^(9\d\d\d\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> <extension required_name="LongDistanceCall"> <condition field="destination_number" expression="^91(\d\d\d\d\d\d\d\d\d\d)$"> <action application="bridge" data="sofia/external/$0(a)rats.marketel"/> </condition> </extension> </context> </section> </document> -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/

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event filter add order
by Anton VG 26 May '11

26 May '11

I am noticed that I do not receive the BACKGROUND_JOB event if filter for the given event is added after filter on Unique-ID filter. e.g. if I do filter Unique-ID XXXXX filter Unique-ID YYYY filter Event-Name BACKGROUND_JOB I do not receive the BACKGROUND_JOB event at all but if I do filter Event-Name BACKGROUND_JOB filter Unique-ID XXXXX filter Unique-ID YYYY I receive it Seems a little odd, since described that it's a filter in Is the given intended behavior or a bug?

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