May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Slow dialplan execution
by Josh M. Patten 09 May '11

09 May '11

I seem to be having a big problem when using the "Mad-Boss" Paging setup described in: http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom To connect 5 phones to the conference bridge takes anywhere from 3 - 4 seconds which is entirely too long to be useful. Here is the dialplan entry: <extension name="Test Paging Group"> <condition field="destination_number" expression="^(3400)$" break="never"> <action application="set" data="api_hangup_hook=conference 3400 kick all"/> <action application="answer"/> <action application="export" data="sip_invite_params=intercom=true"/> <action application="export" data="sip_auto_answer=true"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=none"/> <action application="conference_set_auto_outcall" data="{alert_info=sipXpage}sofia/custom_dialplan/7001@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7002@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7004@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7005@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false,{alert_info=sipXpage}sofia/custom_dialplan/7006@sipxpbx.bc.local;sipx-noroute=VoiceMail;sipx-userforward=false"/> </condition> <condition field="destination_number" expression="^(3400)$"> <action application="conference" data="3400@default"/> <action application="conference" data="3400 kick all"/> </condition> </extension> Here are my questions: Is it possible to speed up XML dialplan execution to overcome this issue? I promise I have enough CPU, RAM, and bandwidth to handle the flurry of SIP messages. If it is not possible to speed up the XML dialplan execution is there a way to "thread" the dialplan execution so that all of these conference_set_auto_outcall applications can be run simultaneously? If not, would attempting to use a database based dialplan be worth my time? Because of the complexity of the networks I'm dealing with I'd rather not try to fight with multicast. Thanks!!

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How to add a new language
by David Notivol 09 May '11

09 May '11

Hi all, I'm trying to make some tests with some custom audio files in English and Spanish, but I'm not being able to add Spanish as language. The module mod_say_es is compiled and loaded. I duplicated the structure in lang/en to lang/es, but without any luck, always getting a "couldn't find es language" message. Any advise or suggestion? Thanks. -- Regards, David Notivol

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Trillium Error
by Dennis Young 09 May '11

09 May '11

All, I'm seeing this error message in the command prompt window that is running Freeswitch but it's not showing up in the freeswitch log file or remote console. I running GIT HEAD 5/1/11 on WIN32 system. UNTSS: sw error: ent: 010 inst: 000 proc id: 000 file: ..\..\trillium\in\in_bdy4.c line: 483 errcode: 14536 errcls: ERRCLS_DEBUG errval: 00018 errdesc: inUsrT302S25() failed, timer not defined for switch. It seem to happen right before in inbound freetdm call rolls to voicemail. No calls in the switch are failing to my knowledge. Any ideas? ...dly Dennis Young |CIO |Phone 512.437.3901 | Fax 512.437.7202 Supportkids Services, Inc. | P.O. Box 18988 | Austin | TX 78760 | Phone 512.437.6000 | Fax 512.437.6030 [cid:image001.gif@01CC0BD9.49D052E0] Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law. Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.

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zrtp error
by Michele Garribba 09 May '11

09 May '11

Hi, I'm trying to compile a FS server with zrtp but i get the following errors cc1: warnings being treated as errors src/switch_rtp.c: In function ‘zrtp_cache_save_callback’: src/switch_rtp.c:644: error: implicit declaration of function ‘zrtp_def_cache_store’ src/switch_rtp.c: In function ‘zrtp_event_callback’: src/switch_rtp.c:686: error: implicit declaration of function ‘zrtp_verified_set’ src/switch_rtp.c:686: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ src/switch_rtp.c:748: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ src/switch_rtp.c:761: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ src/switch_rtp.c:774: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ src/switch_rtp.c: In function ‘switch_rtp_init’: src/switch_rtp.c:828: error: ‘zrtp_config_t’ has no member named ‘def_cache_path’ src/switch_rtp.c:828: error: ‘zrtp_config_t’ has no member named ‘def_cache_path’ src/switch_rtp.c:828: error: ‘zrtp_config_t’ has no member named ‘def_cache_path’ src/switch_rtp.c: In function ‘switch_rtp_zerocopy_read_frame’: src/switch_rtp.c:3476: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ src/switch_rtp.c: In function ‘switch_rtp_write_frame’: src/switch_rtp.c:4068: error: ‘zrtp_session_info_t’ has no member named ‘sas_is_verified’ make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make: *** [all] Error 2 Following the wiki all goes well up to the make step. My server is ubuntu 10.10, base system + FS requirements. libzrtp is libzrtp-0.90.572 Any idea? Wrong versions? Thanks smaikol

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mod_limit Error
by FERNANDO VILLARROEL 09 May '11

09 May '11

Hi All. I rebuild my FS box and now i am getting the following error when i load the module limit: 2011-05-06 23:56:11.229308 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_limit.so **/usr/local/freeswitch/mod/mod_limit.so: undefined symbol: switch_channel_get_variable** How i can solve? Fernando.

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FreeSWITCH, mod_managed and IApiPlugin
by Schenk, Oliver 08 May '11

08 May '11

Hi All, I'm completely new to FreeSWITCH, but in a week I managed to achieve quite a bit. It was quite a learning curve to say the least. This is my current setup: - FreeSWITCH 1.0.7 built from tarball. - Developing on Windows XP with VS2008 in C#. - Configured extension 1024 to connect to a managed dll file, which implements IAppPlugin. - Using X-Lite softphone I can connect to FreeSWITCH as user 1001 and dial 1024 and start going through a menu that I built in C#. It successfully retrieves records from an SQL database and so forth. No problems there. My first step was to create a module that gets called whenever a user dials IN bound. They will hear the menu just described. My struggle now is relating to OUT bound calls. What I want is a module that is started as soon as FreeSWITCH is started and begins executing on an endless processing loop in the background. This will continuously monitor a database and if certain conditions occur an outbound call should be queued and then made. If multiple calls need to be made I guess they will be queued and processed one by one. I can handle the queuing part. At this stage I will be testing using extension 1001 as the receiver of the call using my softphone. Question 1: I've been trying to use a class that implements IApiPlugin, but how do I get it to start when FreeSWITCH starts? As I said it should simply be a never ending thread as long as FreeSWITCH is running. I can't find any information regarding how to "Execute" a managed module immediately when FreeSWITCH has started. Question 2: If IApiPlugin can do this, how do I get the session object? Like this? ManagedSession session = new ManagedSession(); session.Originate(???); I can't find any help at all on this. Thanks very much! Oliver Schenk NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.

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Gateways ACL Context
by FERNANDO VILLARROEL 08 May '11

08 May '11

Dear All. How i can change to a private context like "my_context" for inbound traffic from gateways authorized on ACL and how i can assign different private context for each gateway authorized in ACL, like: <list name="domains" default="deny"> <node type="allow" cidr="190.XXX.XX.XXX/32"/> context foo <node type="allow" cidr="66.X.XXX.XX/32"/> context foo1 </list> Regards.

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freeswitch configure file empty
by Michele Garribba 07 May '11

07 May '11

Hi, I'm tryin to setup a system with freeswitch and zrtp. Using ubuntu server 10.10 Everything works during the install process following the wiki but when i got to the ./configure --enable-zrtp command nothing happens. Looking the configure file, it's empty! I downloaded freeswitch with git, as the guide told. Any suggetion? My i try with wget and pickup the tar? In the zrtp wiki says to use only git to do this. Thanks smaikol

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Routing based on alias
by Joseph L. Casale 07 May '11

07 May '11

I assumed one could route a call to a ua via its alias in addition to its actual domain, but that is not working for me. I have a sofia profile setup as an ip in the subnet it services and its aliased to a more descriptive name, a dialplan tries to bridge a call to 1000@<ipaddr> which works as expected, but when it attempts to bridge to the alias, 1000@foo, it fails? Thanks, jlc

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FS as outbound conference server
by fieldpeak 07 May '11

07 May '11

Hi Friends, i need configure FS as outbound conference server, and control FS to realize it by php webpages and DTMF. if there any open source php webpages will be much better, thanks for any hints. Regards, Charles

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