March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

XML parser bug
by David Villasmil 03 Apr '11

03 Apr '11

Hello, I noticed the following: I have my sofia.conf.xml like this: <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/> </global_settings>  </configuration> when I start FS, latest GIT: freeswitch -version FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500) I get the following output: ./freeswitch -waste WARNING: Wasting up to 8 megs of memory per thread. 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing Engine. 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event dispatch thread 0 Cannot Initialize [[error near line 1521]: unclosed  <param name="debug-presence" value="0"/> </global_settings>  </configuration> Please note the absence of: <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" /> FS Starts normally! Is this the correct behaviour? Isn't comments supposed NOT to be read? Thanks all. David

5 7

Why FS rewrites From header?
by Serge S. Yuriev 02 Apr '11

02 Apr '11

Hello, Please see highlighted lines - why on B-leg From trashed? Is my misconfiguration? Log http://pastebin.freeswitch.org/15937 Configs http://pastebin.freeswitch.org/15941 http://pastebin.freeswitch.org/15939 http://pastebin.freeswitch.org/15940 -- wbr, Serge

5 11

incoming fax calls
by Madovsky 02 Apr '11

02 Apr '11

I'm trying to find a way to dectect a fax or call from the same extension <extension name="myphone"> <condition field="${sip_to_uri}" expression="^(9999)@$${domain}$"> <action inline="true" application="export" data="dialed_extension=$1"/> <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,NORMAL_CLEARING,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/> <action application="set" data="group_confirm_cancel_timeout=1"/> <action application="ring_ready"/> <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 1"/> <action application="export" data="originate_timeout=25"/> <action application="bridge" data="{sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/> </condition> </extension> is there a way to detect a fax before answer (2 rings for example) and avoid phone rings until no answer ? thanks

2 2

Build on solaris fails in libs/esl
by Kim Culhan 01 Apr '11

01 Apr '11

Seeing this on Opensolaris b134 and on latest Solaris 11: making all mod_hash Creating mod_hash_la-mod_hash.lo mkdir .libs Compiling mod_hash.c ... "../../../../src/include/switch_types.h", line 563: warning: integer overflow detected: op "<<" (E_INTEGER_OVERFLOW_DETECTED) cc: Warning: Option -db passed to ld, if ld is invoked, ignored otherwise cc: Warning: Option -ffast-math passed to ld, if ld is invoked, ignored otherwise cc: -W option with unknown program all make[5]: *** [src/esl.o] Error 1 make[4]: *** [/usr/local/src/freeswitch/freeswitch/libs/esl/libesl.so] Error 2 make[3]: *** [mod_hash-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 In freeswitch/libs/esl/Makefile: CXXFLAGS=$(BASE_FLAGS) -Wall -Werror -Wno-unused-variable With mod_hash commented in modules.conf all of fs compiles. Any help is greatly appreciated. -kim --

4 10

Variable interpolation of bridge b leg
by mayamatakeshi 01 Apr '11

01 Apr '11

I am setting channel variable execute_on_answer in my call to application bridge. Like this: <action application='bridge' data='{execute_on_answer=record_session $${base_dir}/storage/recorded_sessions/${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav}sofia/external/ DestinationNumber(a)192.168.2.2' /> The above works, and the application record_session is executed on the leg b. However, the uuid it gets is from leg a, and the timestamp is from the time bridge was executed, which as I understand, is happening because the variable interpolation is performed at the moment the application bridge is executed. So, is there a way to delay variable interpolation to the instant the b leg app is executed? br, takeshi

2 2

"Free" Conference Calling
by Max Clark 01 Apr '11

01 Apr '11

Hello, I've noticed a consistent pattern for SIP Termination providers not completing calls to the "Free" Conference lines due to costs. What's the best way to deal with this? Are there published lists of these providers numbers that I can use to influence my LCR? Are there SIP Termination providers that explicitly deal with these lines? Thanks, Max

5 5

Re: [Freeswitch-users] user and public dialplan
by Madovsky 01 Apr '11

01 Apr '11

forgot to say it's an external call but from inside a cluster and the dialstring is like /sofia/external/9999(a)domain.ltd thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, March 31, 2011 12:33 PM Subject: user and public dialplan example: - 9999 extension exisits in conf/directory - no public dialplan that matches 9999 external call is coming to public dialplan. is FS will consider that 9999 exists or not ? Thanks

3 3

Hangup problem, missing SIP BYE
by dunchan 01 Apr '11

01 Apr '11

Hi! I have a server machine with Freeswitch, and SIP gateway to make outgoung call. The gateway has only IP address authentication. I have own created UA, and if i run it from the server and the called party hangs up, it don't get BYE message. If my UA runs in different machine (other IP), it works well. I have no idea about this. :( I've checked the logs, and compared the two methods, and it seems there are no differnces. I use the newest FS. Any suggestions? thanks, Viktor

2 2

phone callers muted when joining to conference
by deniro 01 Apr '11

01 Apr '11

Hi, When join to the conference through perl program $session->execute("conference",conf-name@conf-profile); somehow phones were getting muted is there any parameters to pass while joining conference that will prevent phone callers be muted automatically thx deniro--

2 1

user and public dialplan
by Madovsky 31 Mar '11

31 Mar '11

example: - 9999 extension exisits in conf/directory - no public dialplan that matches 9999 external call is coming to public dialplan. is FS will consider that 9999 exists or not ? Thanks

2 1