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Wiki documentation (multiple-registrations)
by Luis Jimenez 23 Apr '08

23 Apr '08

Hi, i think there is a bug in the wiky docs, they say you have to specify the multiple-registrations param in sofia.conf, but it does not work that way, in the sample config it is set in sip_profiles/default.xml and that way it works. Luis Jimenez

2 1

voicemail problem
by Pete Kay 23 Apr '08

23 Apr '08

Hi, I am having a voicemail problem: Problem 1 In the wiki, it says that with the below line, the user can check vm without needing to authenticate: <action application="voicemail" data="auth default 1005"/> Instead, it goes to the mailbox asking user to record message: [/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_message.wav] rate 8000hz How come it is not going to the vm? Problem 2 If I change to <action application="voicemail" data="check default 1005"/> it asks me to enter user id follow by #. So, I enter 1005#, and it just hangs up. The curl xml created by fs shows: <document type="freeswitch/xml"> <section name="directory" description="Directory For FreeSwitch"> <domain name=""> <user id=""> <params> <param name="password" value="1234"/> </params> <variables> <variable name="user_context" value="agent"/> </variables> </user> </domain> </section> </document> Since I am using the user 1005 to dial in, so userid and password should work, otherwise, I won't be able to login to dial the number. Any inputs will be greatly appreciated. Regards, Pete

2 2

Segmentation fault (core dumped)
by Luis Jimenez 22 Apr '08

22 Apr '08

If a start FS and reboot a snom that is not previously registered, but is configured, when the phone send the unregister message, FS shows "Segmentation fault (core dumped)" ane exit. it is reporducible. Luis Jimenez

4 6

PDD about 2-3 seconds
by Luis Jimenez 22 Apr '08

22 Apr '08

Hi, i installed FS and configured 2 IP Phones 1000 and 1001, when i call from 1000 to 1001 or backwards, i get too high Post Dial Delay, about 2-3 seconds, any help/sugestion on this? PS: with openser and Asterisk there is not appreciable PDD. Luis Jimenez

3 7

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