April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

redirect in dialplan
by Jonas Gauffin 22 Apr '08

22 Apr '08

Hello I'm trying to get redirect to work in the dialplan. Here's the actual attempt: 13:00:18 [DEBUG] mod_sofia.c:119 sofia_on_ring() sofia/default/anonymous(a)anonymous.invalid SOFIA RING 13:00:18 [DEBUG] switch_core_state_machine.c:65 switch_core_standard_on_ring() Standard RING sofia/default/anonymous(a)anonymous.invalid 13:00:18 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Anonymous->1201@default 13:00:21 [DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions destination_number(1201) =~ /1201/ 13:00:21 [DEBUG] switch_core_state_machine.c:96 switch_core_standard_on_ring() sofia/default/anonymous(a)anonymous.invalid State Change CS_RING -> CS_EXECUTE 13:00:21 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/anonymous(a)anonymous.invalid [BREAK] 13:00:21 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/default/anonymous(a)anonymous.invalid) State RING going to sleep 13:00:21 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/anonymous(a)anonymous.invalid Running State Change CS_EXECUTE 13:00:21 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/default/anonymous(a)anonymous.invalid) State EXECUTE 13:00:21 [DEBUG] mod_sofia.c:156 sofia_on_execute() sofia/default/anonymous(a)anonymous.invalid SOFIA EXECUTE 13:00:21 [DEBUG] switch_core_state_machine.c:128 switch_core_standard_on_execute() Standard EXECUTE 13:00:21 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/anonymous(a)anonymous.invalid Execute set(accountcode=1) 13:00:21 [DEBUG] mod_dptools.c:588 set_function() SET [accountcode]=[1] 13:00:21 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/anonymous(a)anonymous.invalid Execute set(gate_direction=outbound) 13:00:21 [DEBUG] mod_dptools.c:588 set_function() SET [gate_direction]=[outbound] 13:00:21 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/anonymous(a)anonymous.invalid Execute set(gate_context=external) 13:00:21 [DEBUG] mod_dptools.c:588 set_function() SET [gate_context]=[external] 13:00:21 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/anonymous(a)anonymous.invalid Execute set(gate_site_id=1) 13:00:21 [DEBUG] mod_dptools.c:588 set_function() SET [gate_site_id]=[1] 13:00:21 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/anonymous(a)anonymous.invalid Execute redirect(sip:+46706930821@sipgw2.gotanet.se) 13:00:21 [DEBUG] mod_sofia.c:909 sofia_receive_message() Re-directing to sip:+46706930821@sipgw2.gotanet.se 13:00:21 [DEBUG] sofia.c:222 sofia_event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: sofia/default/anonymous(a)anonymous.invalid 13:00:21 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/anonymous(a)anonymous.invalid [BREAK] 13:00:21 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/default/anonymous(a)anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING] 13:00:21 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/anonymous(a)anonymous.invalid [KILL] 13:00:21 [DEBUG] switch_core_session.c:670 switch_core_session_signal_state_change() Kill sofia/default/anonymous(a)anonymous.invalid [BREAK] 13:00:21 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/default/anonymous(a)anonymous.invalid) State EXECUTE going to sleep 13:00:21 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/anonymous(a)anonymous.invalid Running State Change CS_HANGUP 13:00:21 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/anonymous(a)anonymous.invalid) State HANGUP 13:00:21 [DEBUG] mod_sofia.c:255 sofia_on_hangup() Channel sofia/default/anonymous(a)anonymous.invalid hanging up, cause: NORMAL_CLEARING 13:00:21 [ERR] mod_xml_cdr.c:118 my_on_hangup() Error writing [D:\gate\freeswitch\log\xml_cdr\569b8bdf-5a48-9d42-bf21-3db37ab69684.cdr.xml][No such file or directory] 13:00:23 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/anonymous(a)anonymous.invalid, cause: NORMAL_CLEARING 13:00:23 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/anonymous(a)anonymous.invalid) State HANGUP going to sleep 13:00:23 [DEBUG] switch_core_session.c:730 switch_core_session_thread() Session 23 (sofia/default/anonymous(a)anonymous.invalid) Locked, Waiting on external entities 13:00:23 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 23 (sofia/default/anonymous(a)anonymous.invalid) Ended 13:00:23 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/anonymous(a)anonymous.invalid [CS_HANGUP]

3 15

help required to configure a freeswitch client
by srikanth 22 Apr '08

22 Apr '08

Hi all, I am new to this freeswitch, i have started my career recently on voip. i am able to install freeswitch sever in my PC and i am able to connect two xlite clients via this freeswitch server. Now my problem is i want to configure a freeswitch client. please can any one help me out. Regards, Srikanth...

1 0

Called-party state/presence as dialplan condition
by Daniel Hefti 21 Apr '08

21 Apr '08

I know (or at least I think I know) that with sofia, if you try to call a party and they are unavailable, you'll get a busy state back, and you can decide whether to go on from there or not, but if you're not using sofia (in this case, I'm using fifo), is there a way to find out the called party's state prior to actually calling the party to create a condition in the dialplan to act based on that state? I guess what I'm getting at is that I would like to modify the default dialplan such that based on the state of 5900, either pickup or park the call at 5900 so only one number needs to be used to access the called party parked on 5900. Currently the call will show as busy on 5900 and can be transferred using the same number, which is cool, but accessing the lot requires a different number to be called. (That can be kind of hard-to-impossible to do with a single soft key on some/most sip phones.) -Dan

2 3

Gateway auth username
by Mark Campbell-Smith 21 Apr '08

21 Apr '08

Hi! Hopefully an easy question... What is the parameter to define the authorization username for a gateway? <param name="username" value="didnumber"/> <param name="password" value="password"/> I also need something like: <param name="auth-user" value="username"/> I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work.... Thanks! /Mark

5 21

configure errors
by Luis Jimenez 21 Apr '08

21 Apr '08

Ok, i'm dont understand why this error, can somebody help me please? when i run ./bootstrap.sh ./configure --prefix=/opt/freeswitch i get the following error: checking target system type... i686-pc-linux-gnu ./configure: line 1997: syntax error near unexpected token `$PACKAGE_NAME,$PACKAGE_VERSION' ./configure: line 1997: `AM_INIT_AUTOMAKE($PACKAGE_NAME,$PACKAGE_VERSION)' configure: error: ./configure.gnu failed for libs/libsndfile Thanks in advance Luis Jimenez

2 2

libtool imcompatible
by Luis Jimenez 21 Apr '08

21 Apr '08

Hello, when i run the bottstrap after svn, i get the following error, do i have to downrade libtool, or is there any solution? # ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.62 (ok) bootstrap: automake version 1.10.1 (ok) bootstrap: libtool version 2.2.2 found. You need libtool version 1.5.14 or newer installed to build FreeSWITCH from SVN. Thanks in advance. Luis Jimenez

2 2

storing voicemessage and greeting in DB
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I am wondering if storing and retrieving voicemaessage / greeting from DB makes sense especially for a load balancing environment. Is there anyway to do it in freeswitch? If not, is there any way of using some ftp-style storing and retrieving? How is it done typically to have voicemail working under a load balancing situation? Thanks in advance for your input. Thanks, Pete

4 3

Please give suggestion on determining if a DTMF is valid
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I have a need to extra a list of valid DTMF input from DB ( as it may sometimes chang). The way I would do it is to extract the list of valid DTMF from DB before session.answer(). When the caller enters the DTMF when the menu is played, I need to search to see if the input is within the the resultset. There are few ways that I can think of: Option 1: Do a search query whenever the user enter the DTMF to check if it is valide. Option 2: Traverse the resultset (setting cursor to beginning of the resultset) to see if the entered DTMF exists or not. Option 3: Store the entire resultset into a hash table before playing the menu. >From the performance perspective, which one is better? Is there a better way to do it? The problem is that I am already experience some delay when doing the first DB call to get the entire list of valid DTMF from DB. I just think further delay is not good. Is there any good way to search for an element within a resultset? Thank you in advance for your input. Regards, Pete

1 0

Hangup Cause: UNALLOCATED
by Alois Komenda 21 Apr '08

21 Apr '08

Hello, I'm trying to place a call between two SIPp instances over FreeSWITCH. Before the connection is set up completeley FS sends a BYE to the caller. In the logs is see the hangup cause is "UNALLOCATED". That happens only if there is a pause after sending the ACK. Hangup sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [CS_EXECUTE] [UNALLOCATED] 2008-04-17 14:53:46 [DEBUG] switch_channel.c:1293 switch_channel_perform_hangup() Kill sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [KILL] 2008-04-17 14:53:46 [DEBUG] switch_core_session.c:619 switch_core_session_signal_state_change() Kill sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 [BREAK] 2008-04-17 14:53:46 [DEBUG] switch_core_state_machine.c:405 switch_core_session_run() sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 Running State Change CS_HANGUP 2008-04-17 14:53:46 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofe r.de:5060) Calling Handler for State HANGUP 2008-04-17 14:53:46 [DEBUG] mod_sofia.c:258 sofia_on_hangup() Channel sofia/freeswitch.mocon.esk.fraunhofer.de/bob(a)freeswitch.mocon.esk.fraunhofer .de:5060 hanging up, cause: UNALLOCATED If I the call is placed by a softphone the same behavoir occurs. What does that mean? Is it a bug in FS? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

3 6

seek suggestion on TTS option
by Pete Kay 21 Apr '08

21 Apr '08

Hi, I have a question about TTS efficiency. I have some text that I need to read out when someone calls in and these texts do change. Here are two options I am thinking about: In my configuration, I would have DB and FS on different servers. DB server is SCSI. Option 1:Store the text in DB Within JS: 1.Get the text from DB 2. use system (command) to create a wav file on locate file system 3. Use playback to play the wav file Option 2: Pre-created the file using NFS ( at the same machine as DB) Within JS: 1. Get the filename from DB 2. Get the wav file from NFS 3. Use playback to play the wav file In a highly-loaded environment, which option would be preferable considering speed and efficiency? Is there other better option? Any input will be greatly appreciated. Thanks, Pete

2 1