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CHANNEL_CALLSTATE 'switch_channel_perform_set_callstate' - Informations
by David 08 Nov '11

08 Nov '11

Hi, If i create a conference via dialplan tool 'conference' like <action application="conference_set_auto_outcall" data="{sip_auto_answer=false}sofia/internal/220"/> ... <anti-action application="conference" data="555@default <mailto:555@default> 555 TestConf"/> , ... the caller_caller_id_name and caller_caller_id_number in the CHANNEL_CALLSTATE Event for Event calling function 'switch_channel_perform_set_callstate' for the OUTBOUND Channel state 'RINGING' was set to 'FreeSWITCH' and '00000000' I cannot set these variables to another value. No set of channel variables via dialplan tools 'set' or via dialstring variables like '{outbound_caller_id=555}555@default 555 TestConf' or with other variables. How can i set these variables to another value by the dialplan or conference configuration dynamicaly? How can i not ? Gernerally, How can i push more informations to these Event function? Thanks David

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Problem 'adding' video to an audio initiated call, fmtp
by Tyler Winter 07 Nov '11

07 Nov '11

Hi, I'm trying to add video to a non-video initiated call. I should mention that I have no problems initiated and answering a call as video (using Bria softphone and H.264 codec). I captured several test calls to compare the SDP. I found that adding video doesn't work only when initiating the call as audio. In other words, I can initiate the call as video on the A-leg, and answer the call as audio on the B-leg and then add video on the B-leg and video starts working. I think problem is due to there being no fmtp "media attribute" on the 200 OK w/SDP response from FS to the A-leg when I initialize the call as audio, then add video to it. I believe I narrowed down the problem to be the following: - Caller (a-leg) initiated the call as audio. - Callee (b-leg) answered the call as video - Caller (a-leg) "adds" video which sends an INVITE to the switch to negotiate the video codecs - FS responds to Caller (a-leg) with 200 OK (SDP) but the "fmtp" media attribute is missing. On a working video call the fmtp attribute is appended to the SDP along with the rest of the video codec details. Here's an example: this is the INVITEs SDP information to FS from the caller (a-leg) on a working video call (initiated as video): Media Description, name and address (m): video 55752 RTP/AVP 123 124 Bandwidth Information (b): TIAS:1574400 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Media Attribute (a): rtpmap:124 H264/90000 Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 Media Attribute (a): sendrecv 200 OK from FS to the Caller (a-leg): Media Description, name and address (m): video 16454 RTP/AVP 123 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Now, when I try to initiate a call as audio, then add video: INVITE from the caller (a-leg) to FS ("adding" video during call): Media Description, name and address (m): video 64698 RTP/AVP 123 124 Bandwidth Information (b): TIAS:1574400 Media Attribute (a): rtpmap:123 H264/90000 Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 Media Attribute (a): rtpmap:124 H264/90000 Media Attribute (a): fmtp:124 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 Media Attribute (a): sendrecv 200 OK from FS to the caller (a-leg): Media Description, name and address (m): video 23678 RTP/AVP 123 Media Attribute (a): rtpmap:123 H264/90000 So, it appears that the "fmtp" attribute is missing on the 200 OK response: Media Attribute (a): fmtp:123 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 I can upload the captures somewhere if the full details will be helpful to anyone. If anyone has had any luck with this feature, I'm interested in how you accomplished it. I also tried using the setting "sip_force_video_fmtp" to no avail. I appreciate any help. Regards, Tyler

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date/time condition does not work
by Max Machula 07 Nov '11

07 Nov '11

Hello, Hope someone can help me. I have a strange problem with date and time on my FreeSwitch. I have dialplan: <extension name="WW26"> <condition field="destination_number" expression="26"/> <condition field="year" expression="2011"> <action application="bridge" data="USER/26@$${domain}"/> ... skip ... </condition> </extension> Server time: # date Thu Nov 3 21:34:22 GMT 2011 However FreeSwitch can not confirm the year 2011. Log file: Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xx parsing [public->WW26] continue=true Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (PASS) [WW26] destination_number(26) =~ /26/ break=on-false Dialplan: sofia/internal/07711567890(a)xx.xx.xx.xxRegex (FAIL) [WW26] year() =~ /2011/ break=on-false I get the same problem if I change field year to wday which set to 1-6 for example. Any ideas would be much appreciated. Thanks, Max

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HA: Starting FreeSWITCH empties the calls table?
by peely 07 Nov '11

07 Nov '11

Hi, I've been playing about with the HA setup, and have set something up similar to the HA page on the Wiki with two FS boxes using MySQL Master <> Master replication and FreeSWITCH on the secondary box listening on un-mounted IPs. This bit all works very nicely and when both boxes are up I can "fsctl crash" the current active box, then mount the IPs on the soon-to-be-live box then issue "sofia recover", it's all very impressive! What I notice though is that when I bring the "crashed" box back up, starting the freeswitch process causes all calls in the database to be removed which are then replicated across to the active box's database, if I issue a "show calls count" on both boxes the figure goes to 0 immediately after the freeswitch process is started on the secondary. >From what I can see this doesn't affect things on the newly active box, a "status" shows channels in use, much greater than the number of calls, which gradually picks up again. I'm guessing freeswitch issues something like 'delete from calls' when it starts? For HA setups is this an appropriate thing to do? For reference, I'm currently running FreeSWITCH Version 1.0.head (git-fa3fa30 2011-10-07 11-20-36 -0500). Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/HA-Starting-FreeSWITCH-emptie… Sent from the freeswitch-users mailing list archive at Nabble.com.

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one gateway two registrations
by Roman Dissauer 07 Nov '11

07 Nov '11

I have a gateway defined in directory, this works fine but it ties to register over both internal and external sip_profile. How can I restrict it to just the external profile? Thanks! Roman

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Lua time condition
by chrisbware＠interfree.it 07 Nov '11

07 Nov '11

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SIP statistic
by Beppe Grillo 07 Nov '11

07 Nov '11

Hy, I need to retrieve the statistical data of the SIP Protocol. Show me some documentation that you know is this information ? How can I view SIP traffic counters? Thanks, Giuseppe.

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I Have some problems with "sudo make current"
by Павел Бондарь 07 Nov '11

07 Nov '11

I Have some problems with "sudo make current" Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_serialize.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_abyss.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_server_cgi.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_string.c... Compiling ../../../../libs/xmlrpc-c/src/xmlrpc_struct.c... Compiling ../../../../libs/xmlrpc-c/lib/expat/xmltok/xmltok.c... Compiling /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c... quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -fPIC -DPIC -o .libs/mod_xml_rpc.o quiet_libtool: compile: gcc -w -I../../../../libs/xmlrpc-c/lib/expat/xmlparse -I../../../../libs/xmlrpc-c/lib/expat/xmltok -I../../../../libs/xmlrpc-c -I../../../../libs/xmlrpc-c/include -I../../../../libs/xmlrpc-c/lib/abyss/src -I../../../../libs/xmlrpc-c/lib/util/include -D_THREAD -D__EXTENSIONS__ -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c -o mod_xml_rpc.o >/dev/null 2>&1 Creating mod_xml_rpc.la... make[6]: Выход из каталога `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[5]: Выход из каталога `/usr/src/freeswitch/src/mod/xml_int/mod_xml_rpc' make[4]: Выход из каталога `/usr/src/freeswitch/src/mod' make[3]: Выход из каталога `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Ошибка 1 make[2]: Выход из каталога `/usr/src/freeswitch' make[1]: *** [all] Ошибка 2 make[1]: Выход из каталога `/usr/src/freeswitch' make: *** [current] Ошибка 2 How can I fix it????? Ubuntu 10.04 x64

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Problem Verifying an existing file with lua
by Thomas Hoellriegel 07 Nov '11

07 Nov '11

Hi all, I like to export a vatiable to my dialplan. The variable file_name_ok should give back the value true if the file exists My fialplan: <action application="lua" data="file.lua ${call_screen_filename}"/> <action application="log" data="the value: ${file_name_ok}"/> My problem: i can.t export the variable from the luascript my scipt: -- function used to check for the existence of a file function file_exists(fname) local f = io.open(fname, "r") if (f and f:read()) then return true end end if ( session:ready() ) then session:answer( ); datei = session:getVariable("file_name_ok"); if (file_exists(datei..extension..".wav")) then session:setVariable("file_name_ok", "true"); else session:setVariable("file_name_ok", "true"); end end can your help me please? --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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RTP SSRC changes mid session
by Kiss Karoly 07 Nov '11

07 Nov '11

Hello all, I ran into a problem where freeswitch stopped relaying media in bridge mode from leg A to leg B when the endpoint on leg A (Cisco SIP gateway) changed the SSRC in the RTP stream. The first RTP packet with the new SSRC had the Mark bit set. The closest I got when reading up on this issue was RFC3550 Section 8.2, which states that and endpoint may change SSRC but must send an RTCP Bye before doing so. In my case there was no RTCP involved. Am I right to assume that FS stopped relaying media because the SSRC changed ? Is there a magical switch that will tell FS not to stop relaying media in this case ? If there is, are there any caveats ??? Using: FreeSWITCH Version 1.0.head (git-3aeaec7 2011-09-02 23-45-57 +0200) Thanks, Kiss Karoly

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