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Multiple calls with PortAudio
by Jason White 27 Mar '09

27 Mar '09

This has occurred with a number of recent revisions. If anyone can reproduce it or suggest debugging steps, I'll gladly supply more information. Jira isn't convenient for me due to X issues at the moment, unless there's a way to interact with it other than via a Javascript-capable Web browser. Distribution: Debian Sid, now kernel 2.6.29 (same experience with 2.6.26). FreeSwitch revision 12701. Steps to reproduce Make a call with portAudio to a SIP endpoint (I haven't tried it with other endpoints): pa call <destination> Next, make another call: pa call <other_destination> Result: the party to the first call hears no music on hold, or possibly broken/intermittent music on hold, and the second call fails to go through. Where it hangs seems to vary, and sometimes it does in fact work. Often it hangs before it reaches the calling state, suggesting that there's some kind of locking issue involved. I'll perform further testing. Meanwhile, if anyone else can reproduce it, this may help. Note that the second call often doesn't return an error; it just hangs somewhere in the set-up process, e.g., init.

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sipp emulating a registered end point
by Jonas Gauffin 27 Mar '09

27 Mar '09

Hello I want to achive this: Sipp1 -> FS -> Sipp2 Sipp1 emulates a inbound calls (easy to achive) Sipp2 should emulate a registered user (i.e. register with FS and then just wait for calls and hangup when sipp1 hangsup) How do I configure sipp as "Sipp2"? Thanks, Jonas

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Load testing and thread use
by Richard Lamkin 26 Mar '09

26 Mar '09

Dear All, I am testing FS as a call queuing server. My test set up is as follows; Two Windows XP (SP3) Pc's with 1.0.3 installed out of the box. [ Before anyone says use Linux I need to use windows for a specific reason] I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 [extn 1500]. I then use the following CLI command to create a call from FS1[5900] to FS2[5900] "bgapi originate sofia/gateway/fsb1500/5900(a)richardl-5013-2.mettonigroup.com 5900" This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. I have one hardphone on FS2 with a call connected to 5900 to allow me to listen to MOH. I have been monitoring the system resources used by FS and have observed that for each new call a new thread is created. I have pushed well over 200 calls from FS-1 into FS2. All goes well with the CPU usage by FS (<25%) until 150 concurrent calls when it starts to climb rapidly and at 175 (~50%). One good thing is that memory usage is reasonable with 90MB for 150 calls. My questions are; Q1 - Is my test reasonable ? Q2 - Windows XP is not that happy with large numbers of thread per app. Is there a way to configure FS to use fewer threads; instead of using 1/call can it be configured to use to 1/(n calls) ? Q3 - If there's not a way to configure FS to combine threads would it be reasonable to consider modifying the FS code to allow the combining of threads or would this be a mammoth task. Regards Richard Lamkin ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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Action and Anti-Action
by Rodrigo P. Telles 26 Mar '09

26 Mar '09

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Another fifo request
by Matthew Fong 26 Mar '09

26 Mar '09

Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt

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compile poblem - FIXED_POINT or FLOATING_POINT
by Frank ＠ Impact 26 Mar '09

26 Mar '09

I have this version running on fedora 8 right now. Compiled fine and is in production. version FreeSWITCH Version 1.0.trunk (10960) However, I was in the src directory and ran "make current" and after starting to compile it blew up with Making all in libspeex make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c -o nb_celp.lo nb_celp.c gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC -DPIC -o nb_celp.o In file included from modes.h:41, from nb_celp.h:39, from nb_celp.c:37: arch.h:65:2: error: #error You now need to define either FIXED_POINT or FLOATING_POINT make[4]: *** [nb_celp.lo] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/src/freeswitch/libs/speex' make[1]: *** [libs/speex/libspeex/libspeexdsp.la] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 ----- from modules.conf- loggers/mod_console loggers/mod_logfile loggers/mod_syslog applications/mod_commands applications/mod_conference applications/mod_dptools applications/mod_enum applications/mod_fifo applications/mod_voicemail applications/mod_limit applications/mod_expr applications/mod_esf applications/mod_fsv asr_tts/mod_flite asr_tts/mod_pocketsphinx asr_tts/mod_cepstral codecs/mod_g711 codecs/mod_g723_1 codecs/mod_amr codecs/mod_g729 codecs/mod_h26x codecs/mod_voipcodecs codecs/mod_ilbc codecs/mod_speex dialplans/mod_dialplan_xml dialplans/mod_dialplan_asterisk endpoints/mod_iax endpoints/mod_sofia event_handlers/mod_event_socket event_handlers/mod_cdr_csv formats/mod_native_file formats/mod_sndfile formats/mod_local_stream formats/mod_tone_stream languages/mod_spidermonkey languages/mod_spidermonkey_teletone languages/mod_spidermonkey_core_db languages/mod_spidermonkey_socket languages/mod_lua xml_int/mod_xml_rpc xml_int/mod_xml_curl xml_int/mod_xml_cdr say/mod_say_en any insight on what might be doing on with this compile where it previously compiled fine.

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Intermittent startup failures with TLS-enabled profiles
by Jason White 26 Mar '09

26 Mar '09

I have TLS enabled in my internal and internal-ipv6 profiles as per the stock configuration. When FreeSWITCH is started, sometimes either of the profiles fails to initialize, with an "Unable to create SIP UA for profile" error in the log. If I then start the profile manually sofia profile <profile-name> start from fs_cli, the profile starts up as it should. So far, this has only occurred with the profiles for which I have TLS enabled. I can do more testing to see whether that's part of the problem. Meanwhile, has anyone else seen this? It's revision 12701 from svn.

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Compile Errors ...
by Shelby Ramsey 26 Mar '09

26 Mar '09

Hello, I'm getting this all over the place today: make[5]: *** No rule to make target `/usr/src/freeswitch/libfreeswitch.la', needed by `mod_commands.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_commands-all] Error 1 make[2]: *** [all-recursive] Error 1 I normally do this: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk ./bootstrap.sh ./configure make make install Thx! SDR

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Lua session:setInputCallback
by Robert Clayton 26 Mar '09

26 Mar '09

All, When using Lua InputCallback while streaming audio and collecting an undetermined number of digits (finished by #) it seem that returning false, break or stop all close the InputCallback ability. I do not see the difference between the three? My question is in order to collect multiple digits the streaming must continue or be paused. Is there a way to stop the recording in such a way that digits can continue to be collected other than using pause? Bob

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help about the rtp port.
by congxin zhao 26 Mar '09

26 Mar '09

Hi, I meet a issue about the rtp port. The media port value in the sdp of SIP 200 OK which freeswitch internal sends to ua is always 0, so the media forward is always wrong. What could cause this problem possibly? No matter I set <param name="inbound-bypass-media" value="true"/> or <param name="inbound-proxy-media" value="true"/> The problem always showes. Does the following rtp-ip and ext-rtp-ip set correct? In external.xml, I set both rtp-ip and ext-rtp-ip as $${external_rtp_ip}, is that correct? I am not clear about this two variables in the external.xml and internal.xml external.xml: <param name="rtp-ip" value="$${external_rtp_ip}"/> external.xml: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> internal.xml: <param name="rtp-ip" value="$${internal_rtp_ip}"/> internal.xml: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> Thanks, -Congxin

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