June 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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Join Us For A ClueCon Party!
by marketing＠cluecon.com 30 Jun '11

30 Jun '11

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FreeSWITCH Cookbook - New recipes needed
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! I've replaced the old recipe list with a new one. The link is still here on the wiki along with other information for prospective authors: http://wiki.freeswitch.org/wiki/Cookbook#Information_for_prospective_authors We've made a few changes to the content. Specifically, we've removed two chapters: "Call Center Services" and "Carrier Class Services". We do not feel that these can properly be done in "recipe" format. So instead we've take a few of the recipes from those chapters and put them into the "misc." section on the FSCB recipe list. We invite your input on what to replace them with. What do you think? What recipes would you like to see? Send use your ideas. Thanks, Michael

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FreeSWITCH Conf Call - Plivo Discussion - Recordings Up
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! I knew that today's call would be popular so I did a rush job and got the recordings up: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls Just grab one of the three file types from June 29. Enjoy! -MC

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FreeSWITCH Conf Call Today: Plivo, an OSS Twilio Alternative
by Michael Collins 29 Jun '11

29 Jun '11

Hello all! Today's conference call is busy! We have Venky from the Plivo project discussing the new software. We also have quite a few new FS items that have been added in the last month so we need to get some documentation done. :) Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 There is a link in there to a PPT for us to look at during Venky's presentation. Talk to you soon! -Michael

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freeSWITCH pause
by Matthew Margolis 28 Jun '11

28 Jun '11

Hey is there any possible way to find out if the "fsctl pause" command was issued? Like an isPaused call? Matthew Margolis Programmer PATLive T 850.766.3236 E matthew.margolis(a)patlive.com

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patch for switch_ivr_sleep break behavior
by Steve Richardson 27 Jun '11

27 Jun '11

I noticed that the dialplan tools "sleep" function would not respond to "break" if the call did not have ready media (e.g., if it was still ringing and not bridged anywhere). The dptools sleep is just a wrapper call for switch_ivr_sleep(), and I made a small patch for it to look for a break in the case where the media was not ready. I don't know if this has a negative effect on other uses of switch_ivr_sleep() in freeswitch, but for my application this allows me to break a sleeping channel when the media is not ready, allowing it to proceed. Instead of yielding for the full sleep time, it yields in small increments and checks for break after each yield. I chose 20ms, but my application would probably tolerate 100ms without much impact. I don't know if there's a more elegant way to do this, but I'm certainly open to suggestion. Patch follows for your consideration: diff --git a/src/switch_ivr.c b/src/switch_ivr.c index 92eb97e..6c1202a 100644 --- a/src/switch_ivr.c +++ b/src/switch_ivr.c @@ -151,7 +151,15 @@ SWITCH_DECLARE(switch_status_t) switch_ivr_sleep(switch_core_session_t *session, */ if (!switch_channel_media_ready(channel)) { - switch_yield(ms * 1000); + + for (elapsed=0; elapsed<(ms/20); elapsed++) { + if (switch_channel_test_flag(channel, CF_BREAK)) { + switch_channel_clear_flag(channel, CF_BREAK); + return SWITCH_STATUS_BREAK; + } + + switch_yield(20 * 1000); + } return SWITCH_STATUS_SUCCESS; } regards, Steve -- *Stephen Richardson Senior Software Engineer* Comrex Corporation 19 Pine Rd Devens MA 01434 USA v 978.784.1764 f 978.784.1717 steve(a)comrex.com <mailto:steve@comrex.com> http://www.comrex.com/

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