May 2011 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

mod_portaudio multiple endpoints support
by Moises Silva 25 Oct '11

25 Oct '11

Hello, I've been doing some changes to mod_portaudio sponsored by Comrex Corporation. The changes involve adding support for using multiple audio devices simultaneously, including devices with multiple channels (note that is not exactly the same as stereo). The changes are available at git:// github.com/moises-silva/mod_portaudio-endpoints.git Complete documentation on the new functionality in portaudio can be found in the sample configuration in the form of comments (conf/autoload_configs/portaudio.conf.xml). The portaudio changes are totally backwards compatible, existing configurations should keep working as expected. In addition to the new portaudio functionality there are a few other changes that were necessary to use the new functionality. All the changes described below: 1. mod_portaudio.c and pablio.c modifications to allow usage of multiple streams, channels and devices. 2. uuid_outgoing_answer API added to mod_commands.c, this API allows to answer outgoing channels. This may only be useful for portaudio endpoints since most, if not all, endpoints receive the answer notification using some sort of signaling message (200 OK for example). 3. Changes to switch_generate_sln_silence() in switch_resample.c to allow specifying -1 as a special divisor value meaning to zero-out the buffer completely. This option was required by Comrex too since the equipment they have seems to be amplifying the random low-level noise that switch_generate_sln_silence() currently generates by default and they don't like the "hissing noise" on the background while the call is answered and there is no ringback tone. I'd appreciate anyone who can review and comment. I just merged a few moments ago with latest git therefore reviewing should be easy with git diff adding the github repo as a remote. I'd like to get approval from the core FreeSWITCH team to merge this into mainstream. Needless to say I'll be available in the case any bugs are introduced in the portaudio code, virtually impossible, but just in case :-) Thanks, - Moy

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Sipx TLS connections make Freeswitch crash
by barisyanar 11 Aug '11

11 Aug '11

Hi all, I am using Freeswitch 1.0.7 with sipx release-4.4. Whenever I try to make an IVR call with TLS, Freeswitch crashes dumping the attached core file. The output bt command is: #0 __strchr_sse42 () at ../sysdeps/x86_64/multiarch/strchr.S:131 #1 0x00007f9c0b4373c3 in sofia_glue_get_url_from_contact (buf=0x0, to_dup=0 '\000') at sofia_glue.c:4943 #2 0x00007f9c0b438e1b in sofia_overcome_sip_uri_weakness ( session=0x7f9c04017108, uri=<value optimized out>, transport=SOFIA_TRANSPORT_TCP_TLS, uri_only=SWITCH_TRUE, params=0x0) at sofia_glue.c:1296 #3 0x00007f9c0b4182b0 in sofia_handle_sip_i_invite (nua=0x7f9c04008890, profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=<value optimized out>, sip=<value optimized out>, tags=<value optimized out>) at sofia.c:6793 #4 0x00007f9c0b433aca in sofia_event_callback (event=nua_i_invite, status=100, phrase=0x7f9c0000ab40 "Trying", nua=0x7f9c04008890, profile=0x148dad0, nh=0x7f9c00009ec0, sofia_private=0x0, sip=0x7f9c00006658, tags=0x7f9c0000ab30) at sofia.c:942 #5 0x00007f9c0b4a5f53 in nua_application_event (dummy=<value optimized out>, sumsg=<value optimized out>, ee=0x7f9c0000ab08) at nua_stack.c:393 #6 0x00007f9c0b503974 in su_base_port_execute_msgs (queue=0x0) at su_base_port.c:280 #7 0x00007f9c0b503ec1 in su_base_port_step (self=0x7f9c040010f0, tout=0) at su_base_port.c:473 #8 0x00007f9c0b42d4ea in sofia_profile_thread_run ( thread=<value optimized out>, obj=0x148dad0) at sofia.c:1623 #9 0x0000003efd2068e0 in start_thread (arg=0x7f9c18116710) ---Type <return> to continue, or q <return> to quit--- at pthread_create.c:297 #10 0x0000003efcee0c9d in clone () And sipx's freeswitch log prints the line: 2011-05-16 15:04:51.530426 [WARNING] switch_core.c:1120 Cannot locate domain deneme.deneme.karelarge.com, where deneme.deneme.karelarge.com is my fqdn. No problem occurs during TCP connection. Problem is possibly caused by sipx, but i'll be glad to hear FS community's ideas.

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Removing SDP from 183/180
by Artur Cichocki 05 Jul '11

05 Jul '11

Hi All. I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes SDP from 183 and returns 180 (it also removes SDP from 180, because FS changes 180 with SDP to 183). To turn it on add {sip_remove_183sdp=true} to the dial string. Why? Because in some situations we don't want to transfer voice between clients before a real call establishment (and yes, I know that early media in this stage is allowed by RFC). Real case: Try to setup two clients, one of them using Sipdroid. Turn on bypass mode. Try to call to Sipdroid client, you will hear caller voice together with the ring (because Sipdroid sends 180 with SDP, and puts voice on the speaker). Patch attached. -- Artur Cichocki

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mod_socket
by Ali R. 05 Jul '11

05 Jul '11

Hi everyone I think my issue could be fixed with mod_xml_curl but I'm using FS purely through sockets (inbound mode) When I get the park event, I execute some biz logic. Leg's A source IP plays a role in this logic so based on the value of the IP (a big pool of allowed IP addresses is generated dynamically and changes very fast so I have a logic to query this pool) all good so far. However, if the incoming channel’s source IP is not allowed I would still want to answer and continue on but I must challenge it with a user name and password that I have a logic to retrieve. Any thoughts on how should I go about doing the username/password challenge through the socket without starting a new listening socket for mod_xml_curl requests . I really appreciate any thoughts on this issue Many Thanks,

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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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