October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch - mod_opal, mod_h323 never get installed
by Shamun toha md 05 Oct '10

05 Oct '10

Hello, Fedora 12 - mod_opal or mod_h323 not able to install. Here is the following: Installed: opal-devel.i686 0:3.6.8-1.fc12 Dependency Installed: ptlib-devel.i686 0:2.6.7-1.fc12 speex-devel.i686 0:1.2-0.12.rc1.fc12 [root@example freeswitch]# make mod_opal making all mod_opal Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall&, FSEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)’ /usr/include/opal/opal/localep.h:249: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*) /usr/include/opal/opal/localep.h:242: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function ‘switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1296: error: ‘class OpalMediaPatch’ has no member named ‘OnStartMediaPatch’ /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function ‘switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1418: error: ‘class OpalMediaPatch’ has no member named ‘OnStartMediaPatch’ make[4]: *** [mod_opal.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opal-all] Error 1 make[1]: *** [mod_opal] Error 2 make: *** [mod_opal] Error 2 [root@example freeswitch]# Thanks

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test
by Holger Esser 05 Oct '10

05 Oct '10

test

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Play_fsv application
by Joshua Foshee 05 Oct '10

05 Oct '10

I have something strange happen when I play back a fsv file. It will play super-fast. You can hear the audio go fast too then you get a long busy. Any ideas what would cause the speed to increase on playback? Thanks, Josh

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FreeSWITCH overrides/dose no accept hangup cause
by ivdreg ivdreg 05 Oct '10

05 Oct '10

Hi All, I have 2 FS boxes that communicates each other. First originates call to second and on second FS perl scrips process a call. On some errors during processing I hangup call with specific hangup cause and everything is OK. For example: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 10.10.1.101;rport=5060;branch=z9hG4bK77698B1e1BZ3j From: "11111111111" <sip:11111111111@10.10.1.101<sip%3A11111111111(a)10.10.1.101> >;tag=X628H5HUDy86S To: <sip:999999999999@10.10.1.110:7060>;tag=Ztg1r5S60rQFF Call-ID: cca7076d-4b12-122e-44b6-7fd0651f7bfe CSeq: 2802191 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer *Reason: Q.850;cause=38;text="NETWORK_OUT_OF_ORDER"* Content-Length: 0 Remote-Party-ID: "999999999999" <sip:999999999999@10.10.1.110<sip%3A999999999999(a)10.10.1.110> >;party=calling;privacy=off;screen=no As you can see Q.850 cause is 38 - NETWORK OUT OF ORDER but on first FS application info says: variable_proto_specific_hangup_cause: [sip:503] variable_sip_hangup_phrase: [Service Unavailable] variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE] SIP Cause of both NORMAL_TEMPORARY_FAILURE and NETWORK_OUT_OF_ORDER is 503 and is the same but Q.850 cause is different. Is it normal behavior for 1.0.6 ? Is it also possible first FS to respect Q.850 cause ? Thanks

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Caller ID lost on second bridge in extension
by Mario 05 Oct '10

05 Oct '10

I have a 15 second ring to local extensions, then 15 seconds to local + 2 cell phones. However, when the second bridge is executed the caller ID is replaced by 0000000000. I tried removing pieces below like the execute app but it made no difference. Not found on the wiki or web either. Anyone know where/why it's lost and how to keep it? Thanks. Mario <extension name="u1phone"> <condition field="destination_number" expression="^2920$">  <action application="set" data="ringback=${us-ring}"/>  <action application="execute_extension" data="2921 XML default"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>  <action application="set" data="call_timeout=15"/> <action application="bridge" data="group/home@${domain_name}"/>  <action application="execute_extension" data="2922 XML default"/> <action application="playback" data="/usr/local/freeswitch/conf/mvsounds/we_trying_please_hold.wav"/>  <action application="set" data="ringback=$${hold_music}"/>  <action application="set" data="ignore_early_media=true"/>     <action application="bridge" data="<originate_timeout=15>group/home@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>    <action application="answer"/> <action application="voicemail" data="default ${domain_name} 100"/> </condition> </extension>

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RTCP support
by Dave Redmore 05 Oct '10

05 Oct '10

Hi All, I've done some investigating into getting RTCP support working, but am finding a lot of conflicting information. The WIKI says RTCP can be configured on a per session bases or for an entire profile. I've added "<param name="rtcp-audio-interval-msec" value="1000"/> to my profile config and restarted FS, but am not seeing any RTCP packets in my captures. Is there something else that needs to be done? Thanks. Dave Redmore Spigot Networks, Inc.

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api_on_answer
by Bernhard Suttner 05 Oct '10

05 Oct '10

Hi, I use api_on_answer to call a lua script. Within the lua script I want access to session variables but the lua variable "session" does not exists. How could I get access to them? Thanks in advance. Best regards, Bernhard

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No autoanswer with Originate call
by Eli Hayun 05 Oct '10

05 Oct '10

Hi When I am dialing from A phone to B phone and I do : <action application="set" data="sip_auto_answer=true" /> <action application="set" data="sip_h_Call-Info=<sip: ${domain_name}>;answer-after=2" /> everything is OK When I try from lua script api:executeString("originate [originate string] AUTO_ANS xml default") and in the dialplan I put the set autoanswer=true it is not working. any help? Eli

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Are there mandatory extensions in any context?
by covici＠ccs.covici.com 05 Oct '10

05 Oct '10

Hi. I notice that there are a few extensions at the beginning of the default context in the dialplan such as unroll-loops and a few more. I wonder if I make a new context for some specific purpose, to hold conferences, for instance -- do I need such extensions in that context as well? thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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High Availability and hostname used by Sofia
by Leon de Rooij 05 Oct '10

05 Oct '10

Hello List, This is going to be a long mail, please bear with me :-) I'd like to configure High Availability for a FreeSWITCH installation by using a failover server. I already discussed it a bit with Sekil on IRC, but I thought it'd be useful to discuss it here as well. The idea is as follows: - I have 2 servers both with an identical FreeSWITCH installation, FS1 and FS2. - Both servers have different static IP's but also one floating IP by means of heartbeat / haresources. - Both FS instances have one Sofia SIP profile that use the floating IP. - Both FS instances are always started (so the backup server will fail to load its SIP profile because the IP is not configured - which is expected behaviour) - There'll be a /etc/ha.d/resource.d "resource script" that starts the SIP profile (through fs_cli -x) the moment it becomes primary - The resource script will also send a "sofia recover" after the SIP profile has been brought up This way: - There'll be one logical switch. - The moment FS2 becomes primary, users will be reachable through the entry in 'sip_registrations' table in the ODBC db, even though they may have registered on FS1 - FS2 will start to send SIP options to NAT'ed useragents (keepalive / udp hole punching) the moment the SIP profile is started. - No client useragent involvement is necessary when failing over (like re-register) - Running calls will be continued because of the "sofia recover" Only problem here is that when a useragent registers to us, then FS also stores its own hostname in "sip_registrations" table in the columns "hostname" and "orig_hostname" (from mod_sofia_globals.hostname). This column is also used in the SELECT query when bridging to a user - so that means if the user registered to FS1, then a failover happens, then if a call is bridged towards that user on FS2, it won't find it because it will search with FS2's hostname in the SELECT query, right ? If everything I'm writing so far is correct, then would it be alright to override the hostname in sofia.conf ? I think that would solve my problem. I already looked it up - it is first set in mod_sofia.c at line 4565 through gethostname(). It can be overriden in sofia.c (function config_sofia) around line 2845. It would probably also work to give both hosts an identical hostname in the OS, but I don't like that (because of rsyslog and more). I'm not convinced yet on how to proceed, please let me know if I'm going the wrong direction. If I'm successful on getting this running, I'll document everything on the wiki - also the haresources script I already wrote I'll put in my contrib dir (or elsewhere in main tree if preferred). Thanks for your time, Leon

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