August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

custom avp in radius
by Hristo Trendev 20 Aug '09

20 Aug '09

Hi, Does anyone know of a way to add custom information in AVP in radius accounting messages? I can see in the radius dictionary there are many AVPs defined, but I'd like to add some additional custom text information. "Freeswitch-AVPair" seems like a generic AVP, but I cannot figure out how to use it (if it's intended for something like that at all). Maybe there is a way to set some variable in the dial plan and to use its contents to populate a general purpose AVP? If there is no such functionality what will be the best way to do it? I found similar feature added in the past (http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=7529) and it doesn't seem too complicated. BR

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Passing Caller ID in Junction Network
by Klaus Teller 20 Aug '09

20 Aug '09

Hi, Anybody with experience in passing caller ID number in junction network? I need some help. I have the following origination string: {origination_caller_id_number=1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/1514xxxxxx Unfortunately, with this, junctionetwork won't pass the caller ID. Any idea? Thanks, Klaus. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser

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No Dialplan on answered channel, changing state to HANGUP
by Max Bridgewater 19 Aug '09

19 Aug '09

Hi, I'm trying to use les.net for termination. When i place a call from the command line, i get the following error message: No Dialplan on answered channel, changing state to HANGUP. What does it mean? My dialing string is as simple as it could be: sofia/gateway/ did.voip.les.net/13028838864 & park() My les.net profile is located under /usr/loca/freeswitch/conf/sip_profiles/external <include> <gateway name="did.voip.les.net"> <param name="username" value="1690xxxxxx"/> <param name="from-user" value="1690xxxxxx"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="ma34n2th./an4"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> Regards, Max.

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playback to legB before bridge?
by Max Ivanov 19 Aug '09

19 Aug '09

Hi all! Is it possible to execute playback application to legB before bridge? I mean sequence of actions similar to this: 1. originate legB 2. playback 3. bridge legA with legB using existing session from step 1 (no new incoming call)

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FreeSWITCH Console Released For iPhone/iPod Touch Users
by Chris Danielson 19 Aug '09

19 Aug '09

Announcing the release of FreeSWITCH Console in the Apple Application Store. The application is FREE and allows you to connect to a FreeSWITCH event socket layer module that is bound to an external interface. Great for development purposes and general remote debugging. Blog announcement: http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitc… iTunes Store Link: http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221… <http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221…> Kind Regards, Chris Danielson

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how to set different action for different cause code
by Woody Dickson 19 Aug '09

19 Aug '09

Hello, I find hangup_hook, but I would like to define different actions for different hangup codes. Is there anyway to do that? Woody

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Conference Status
by Alan Chandler 19 Aug '09

19 Aug '09

In the command to lists a conference it talks about member status, shown as hear|speak|floor What does the floor mean as a status - I can't find any reference to it anywhere. -- Alan Chandler http://www.chandlerfamily.org.uk

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Stuck on my first attempt at dialplanning
by Alan Chandler 18 Aug '09

18 Aug '09

This is my first attempt at setting up the dial plan to do anything more than the basic default stuff. What I am trying to do is set up some conferencing as follows 1. A calls B 2. B puts both A and B into a conference 3. A or B calls out from the conference (via the * set in caller controls) to C 4a If C hangs up or rejects the call, A/B should go back to the conference 4b If A/B types *1 before the call is answered he should go back to the conference 4c If A/B types *1 after the call is answered both A/B and C should go back into the conference. It nearly all works except i) A seemingly can't use the * caller control, where as B can. The other caller control I set up (mute) seems to work ii) If C rejects the call before answering, B gets hung_up (because of i) I can only test with B) iii) When B gets back to the conference, the * caller control doesn't work (although the mute does) Can someone tell me where I am going wrong. ---------------------- Here is the dialplan snippet that I use to handle the dialout from the conference (pressing the * in step 3)  <extension name="cco"> <condition field="destination_number" expression="^cco$"> <action application="read" data="3 3 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="bind_meta_app" data="2 a s execute_extension::conf_transfer XML features" /> <action application="set" data="hangup_after_bridge=false" /> <action application="bridge" data="user/${digits}@${domain_name}" /> <action application="info" /> <action application="conference" data="${conference_name}@default" /> </condition> </extension> <extension name="conf_transfer"> <condition field="destination_number" expression="^conf_transfer$" > <action application="answer" /> <action application="transfer" data="-both ${conference_name} XML default" /> </condition> </extension> And here is the bit in my default XML where I define the conference <extension name="nb_conferences"> <condition field="destination_number" expression="^(2(0[1-9]|[1-9][0-9]))$"> <action application="answer"/> <action application="conference" data="$1@default"/> </condition> </extension> The call out to the conference from the original call is done via changing the bind meta app in the default dialplan for local extension <action application="bind_meta_app" data="2 ab s execute_extension::cf XML features"/> and this extension in features  <extension name="cf"> <condition field="destination_number" expression="^cf$"> <action application="answer"/> <action application="transfer" data="-both 2${dialed_extension:1} XML default"/> </condition> </extension> I can get as far as talking to C after dialing him, but if he doesn't answer, or rejects the call I get lost in space somewhere. If I type *1 during the call with C I get back to the conference, and C appears to be in the conference (hear him speak) but he gets music on hold. What am I doing wrong? -- Alan Chandler http://www.chandlerfamily.org.uk

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failover based on initial INVITE timeout
by Hristo Trendev 18 Aug '09

18 Aug '09

I am trying to implement failover dialing plan as described in: http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout I figured out that originate_timeout must be passed as {originate_timeout=<timeout>} in front of the dial string to have any effect (setting it as channel variable as described in the example above has no effect). I have set the timeout to 1 second, so expected behavior is to try the second gateway if no response is received from the first one in 1 second. The problem is that FS cancels the first request with [NO_ANSWER] and tries to route the call via the second gateway even though it receives response from the first during that 1 second. The response received is "100 Trying" provisional response (checked with sofia siptrace). I'm guessing that either 100 provisional responses don't cancel the originate_timeout timer (bug?) or I am doing it the wrong way. I was also thinking of using the timer-T1 or timer-T1X64 parameter in the sip profile, but I need this to be set per dial string, not per profile, besides, it seems that these timers (T1, T1X64) affect both invite and non-invite requests, so this is not really an option. Also, I tried leg_timeout, but it doesn't really do what I need it to. Anyone has any idea how to implement this?

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memory leak with switch_core_hash_insert
by Juan Backson 18 Aug '09

18 Aug '09

Hi, I am getting some strange vg malloc error message in switch_core_hash_insert. Does anyone know what is wrong with these few lines? Am I missing something? switch_core_hash_init(&hash,pool); param_name =switch_core_sprintf(pool,"%s", key); param_value =switch_core_sprintf(pool,"%s", value); switch_core_hash_insert(hash,param_name,param_value); ==5459== ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are definitely lost in loss record 62 of 89 ==5459== at 0x4A05809: malloc (vg_replace_malloc.c:149) ==5459== by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278) ==5459== by 0x4CE1C7E: rehash (hash.c:227) ==5459== by 0x4CE206F: sqlite3HashInsert (hash.c:386) ==5459== by 0x4C70EA4: switch_core_hash_insert (switch_core_hash.c:67) ==5459== by 0x7CEAD07: ??? (mod_specialivr.c:343) ==5459== by 0x7CEBF87: ??? (mod_specialivr.c:784) ==5459== by 0x7CEDBF0: ??? (mod_specialivr.c:1778) ==5459== by 0x4C790AC: switch_core_session_run (switch_core_state_machine.c:109) ==5459== by 0x4C751FF: switch_core_session_thread (switch_core_session.c:1066) ==5459== by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so) ==5459== by 0x35018D1DEC: clone (in /lib64/libc-2.5.so) ==5459==

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