December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Proto specific hangup cause issue
by Apostolos Pantsiopoulos 09 Dec '08

09 Dec '08

The proto_specific_hangup_cause is missing on one of the two call legs. When the caller hangs up it is missing from the a-leg CDR. When the callee hangs up it is missing from the b-leg CDR. Is this nornal? And another question : what piece of info could inform me about who hanged up? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

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Catching hangups
by Joe Bain 09 Dec '08

09 Dec '08

Hi, I'm writing an IVR in Lua and am having problems dealing with hangups cleanly. Very often session:ready() reports true long after I have hung up and the hangup hook function I have set doesn't get called either. It seems to report that the session is active indefinitely in some cases where a loop keeps trying to get some dtmf key presses. Is there any trick to using session:ready() or the hangup hook that I might have missed? On a slightly related point I can't seem to access the session properties, e.g. session.caller_id_num has a value of nil. Any thoughts here? Thanks in advance, Joe Bain

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root privs for mod_fax
by Gabriel Kuri 09 Dec '08

09 Dec '08

I've been experimenting with mod_fax and discovered it doesn't appear to receive faxes unless freeswitch is running as root? it fails trying to open the tiff file for writing (see the logs below). I'm using the dialplan as prescribed in the wiki without any changes and the user the freeswitch process is running under has privs to write to /tmp, but it still fails to receive faxes. I haven't tried sending any faxes yet. I'm running r10609. any ideas? 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Get document at 14400bps, modem 7 2008-12-09 00:29:41 [WARNING] mod_fax.c:133 spanfax_log_message() WARNING T.30 Cannot open target TIFF file 'rxfax.tiff' 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from state 17 to 3 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Tx: DCN with final frame tag 2008-12-09 00:29:41 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Tx: ff 13 fa 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 HDLC carrier down in state 3 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D _TX 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set rx type 0 2008-12-09 00:29:42 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set tx type 4 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Disconnecting 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set rx type 0 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW FAX Set tx type 1 2008-12-09 00:29:43 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Changing from state 3 to 2 2008-12-09 00:29:44 [DEBUG] mod_fax.c:133 spanfax_log_message() FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2008-12-09 00:29:44 [DEBUG] mod_fax.c:163 phase_e_handler() =============================================================== =============== 2008-12-09 00:29:44 [DEBUG] mod_fax.c:176 phase_e_handler() Fax processing not successful - result (41) TIFF/F file cannot be opened. Gabe

2 2

Two major flaws: Could they be fixed?
by Dennis 09 Dec '08

09 Dec '08

hi, we are fighting with two flaws in fs and would be happy, if they could be fixed. we are using socket outbound. 1.) hangup a call in ringing state: this worked in one of the last fs versions, but suddenly does not work anymore. let's say, we have an inbound call and do 3 originates to different targets. all 3 targets are in ringing state. the target, which answers first, will be bridged with the inbound call, the other two (still ringing) targets should be hung up. we do not want fs to hang up the other two originates automatically. we want to hang up the other two originates by sending the hangups. we set "hangup_after_bridge=false" and "park_after_bridge=true". we do sendmsg hangup uuid. but the originates are first hung up, when they are answered. when they are in ringing state, the hangup will do nothing (anymore). as i said, it worked before, so i assume, that something has changed in the latest trunks. 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid: let's say, we have an inbound call and an outbound call - at least we thing we have it ;-). now we do for example sendmsg outbound_uuid hangup to hangup the outbound call. but, if the uuid of the outbound call does not exist (because there was a problem or something), the inbound will be hung up instead. the same happens with all sent messages to an uuid, which does not exist. if we want to do a playback for the same outbound, the inbound will hear it, if the outbound_uuid does not exist. perhaps this is a feature, but i think that it would be nicer and more reliable, if the sendmsg is only executed on the given uuid. if the given uuid does not exist, nothing should happen or even nicer, an event with an error should be sent to the socket. thanks dennis

4 11

Wrong # in voicemail
by ccav 09 Dec '08

09 Dec '08

My dialplan is pretty simple. I have a single trunk with a vonage softphone DID (1303... we'll call it main) and a "virtual" DID (1816...) which rings the softphone DID. All incoming calls show up as from softphone DID but the sip_to_user holds the actual number dialed so I can enter the dialplan properly. I have 2 extensions in my directory/extensions, one for each of the DID's. The extensions check sip_to_user for match and that works great. I match on ([0,1]?)(<10 digit did>) and it enters the dialplans correctly, plays the right music for each DID while the dial is occuring, so all that works. The bridge to user/$2@$${domain} also works fine. The continue_on_fail is set properly so on no answer call_timeout hits (at 25 secs), and goes to voicemail... works also for both numbers. transfer to voicemail is as follows <action application="answer"/> <action application="voicemail" data="default $${domain} $2"/> which should be pulling $2 from the condition check shown above, which it does, cuz the bridge works... When I call in on main DID, I get "leave a message for 1303..." The Main DID.. When I call in on virtual, I get "leave a message for 1303..." The Main DID rather than the 1816.... How can I get voicemail to use the correct DID. HELP!! :D -- View this message in context: http://www.nabble.com/Wrong---in-voicemail-tp20791453p20791453.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

9 28

mod_xml_ldap
by Helmut Kuper 09 Dec '08

09 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried compile FS with mod_xml_ldap with trunk of yesterday. During compiling it can't find http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz. I looked there and found that the filename on freeswitch.org side has changed to http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tar.gz regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkk+RHwACgkQ4tZeNddg3dyp7wCeN7fvIj4OSH1rsuglD46qtS36 iR8AnjypwB2XT/rAYr61yyMXJ+iUY4/d =iJw/ -----END PGP SIGNATURE-----

3 2

Read app ignores custom variable when called via socket interface
by Jan Kubr 09 Dec '08

09 Dec '08

Hi, I checked out the current trunk (rev 10641) and found out that the read app ignores the varname parameter, it always puts the result in the DTMF-Digit variable. I'm calling it via the socket interface: sendmsg call-command: execute execute-app-name: read execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 10000 # event-lock:true In the XML dialplan it works fine: <action application="read" data="1 1 sounds/en/us/callie/conference/8000/conf-pin.wav res 10000 #"/> <action application="phrase" data="spell,${res}"/> I have been using the socket call above successfully in the 1.0.1 release. Any ideas? Thanks, Jan Kubr

2 4

No audio after transfer
by John Rutherford 08 Dec '08

08 Dec '08

I'm trying to get an attended transfer work with freeSWITCH, but it's not quite working. I have Microsoft Speech Server on one side and Televantage on the other. MSS is originating a call, which freeSWITCH is bridging to Televantage. That calls connects just fine. Then, MSS sends a re-INVITE to Televantage to put the call on hold. This works too. Then, MSS originates another call to freeSWITCH, which is again bridged to Televantage. This works fine too. Then, MSS sends a REFER to freeSWITCH to do the transfer. The transfer should be complete, but there is no audio between the two calls-just silence. I have looked at pcaps and the freeSWITCH logs, but I'm not seeing anything obviously wrong. After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John

2 2

Two nice to have features in fs
by Dennis 08 Dec '08

08 Dec '08

hi, we are using socket outbound and found out, that there are two (perhaps) small things, we would like to see in fs. 1.) if we try to playback a soundfile, which does not exist, we do not get an error or something. in the cli of fs we can see the error, but we do not get anything about it over the socket. we get an execute event for the playback and directly after it an execute-complete. wouldn't it be nicer, if one could get a notice about an error, like: execute-complete-error or a variable, which is set with an error, if an error occours? 2.) the session heartbeat event is an absolutely great feature in fs. but sometimes we would like to get the session heartbeat every 5 seconds. at the moment the allowed minimum is every 10 seconds. a smaller setting, like every 5 seconds, will result in a session heartbeat of every 60 seconds. to help us out, we edit the switch_core_session.c at line 899 and change the "seconds < 10" to "seconds < 5". because one has to set "enable_heartbeat_events=5" manually, i do not think, that there is a risk, that others, who do not want the heartbeat to come that often, will be negatively affected by this change. might it be possible, to do the same changes to the default code? thanks dennis

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rootkit?
by Martin Joseph 08 Dec '08

08 Dec '08

What is the rootkit item that appears to be added to the SVN of trunk? Thanks, Marty

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